Data transfer

ABSTRACT

This application relates to methods and apparatus for transfer of multiple digital data streams, especially of digital audio data over a single communications link such as a single wire. The application describes audio interface circuitry comprising a pulse-length-modulation (PLM) modulator ( 204 ). The PLM is responsive to a plurality of data streams (PDM-R, PDM-L), to generate a series of data pulses (PLM) with a single data pulse having a rising and falling edge in each of a plurality of transfer periods defined by a first clock signal (TCLK). The timing of the rising and falling edge of each data pulse is dependent upon on a combination of the then current data samples from the plurality of data streams. The duration and position of the data pulse in the transfer window in effect defines a data symbol encoding the data. Circuitry for receiving and extracting the data is also disclosed. An interface receives the stream of data pulses (PLM) and data extraction circuitry ( 202 ) samples the data pulse to determine which of the possible data symbols the pulse represents and determines a data value for at least one received data stream.

This application is a continuation of U.S. Non-Provisional applicationSer. No. 13/715,495 filed on Dec. 14, 2012, issuing as U.S. Pat. No.9,424,849 on Aug. 23, 2016, which claims the benefit of U.S. ProvisionalApplication No. 61/570,591 filed on Dec. 14, 2011, U.S. ProvisionalApplication No. 61/639,287 filed on Apr. 27, 2012, U.S. ProvisionalApplication No. 61/639,450 filed on Apr. 27, 2012, U.S. ProvisionalApplication No. 61/673,635 filed on Jul. 19, 2012, U.S. ProvisionalApplication No. 61/681,041 filed on Aug. 8, 2012, U.S. ProvisionalApplication No. 61/695,228 filed on Aug. 30, 2012, United Kingdom PatentApplication No. GB1121524.1 filed on Dec. 14, 2011, United KingdomPatent Application No. GB1207379.7 filed on Apr. 27, 2012, UnitedKingdom Patent Application No. GB1207387.0 filed on Apr. 27, 2012, andUnited Kingdom Patent Application No. GB1215487.8 filed on Aug. 30,2012, all of which are incorporated by reference herein in theirentirety.

FIELD OF DISCLOSURE

This application relates to methods and apparatus for data transfer,especially to the transfer of multiple channels of digital audio data,and in particular to audio data transfer that can use a singlecommunications channel.

BACKGROUND

In many electronic devices there is a need to transmit audio datasignals, or other data signals, within the electronic devices, or tosome peripheral or accessory device(s) that may be attached to theelectronic devices, for instance a set of headphones. In many modernelectronic devices, especially devices with RF transmission capabilitysuch as a mobile telephone or tablet computing device, analogue audiosignals are liable to be corrupted by electromagnetic interference (EMI)and coupling from other nearby circuitry. It is therefore desirable totransmit signals from an audio processing device, e.g. an integratedcircuit audio hub/codec, to the driver of a transducer e.g. speaker, ina digital format so as to preserve the integrity and quality of theaudio signal. For audio data there is also typically a requirement tosend data for multiple channels, for instance left (L) and right (R)data channels for stereo audio.

Modern electronic devices such as smartphones, tablets and the liketypically transmit digital audio data using three data wires plus aground wire. The signal data for both L/R audio channels is typicallysent on one wire in serial format, for example in words of 24 bits, withseparate words being sent for the left (L) data channel and the right(R) data channel. The bit clock signal (BCLK) is sent on another wireand a further clock at the left/right words rate (LRCLK) sent on afurther wire as illustrated in FIG. 1.

Alternatively, the audio data may be transmitted as a high-rate 1-bitstream, sometimes referred to as Pulse Density Modulation (PDM).Word-length reduction, noise shaping and/or delta-sigma techniques maybe applied to reduce quantisation noise in the audio band at the expenseof noise at higher frequencies so as to reduce the required bit rate.However, this still requires two wires for the transfer of digital audiodata, one wire for the data and one wire for the clock. Moreover, thedigital data spectrum will include components corresponding to the baseband audio signals, which may couple onto other analogue linesassociated with the device. Stereo data may be transmitted along onewire, typically alternating between sending left channel data and rightchannel data, but the separate clock line is still needed and thedigital data spectrum may still present similar problems.

SUMMARY

It is therefore an object of the present invention to provide methodsand apparatus for data transfer of multiple data channels that at leastmitigate some of the aforementioned disadvantages.

Thus according to the present invention there is provided digital datatransmission circuitry comprising:

-   -   at least two data inputs for receiving respective input digital        data streams of data bits; and    -   a pulse generator, responsive to said input digital data streams        and to a first clock signal, to generate a single data pulse        having a rising edge and a falling edge within each of a        plurality of transfer periods defined by said first clock        signal,    -   wherein the time of occurrence of the rising edge and the        falling edge of the data pulse encodes the then current data        bits of said input digital data streams.

The occurrence said rising and falling edges of the data pulse arequantised in time within the transfer period. In this way each datapulse corresponds to one of a set of possible data symbols and the datais reversibly encoded, i.e. it may be recovered by a suitable decoder.The rising and falling edges of each data pulse may advantageously besynchronised to a second clock signal, where the second clock signal hasa frequency greater than the first clock signal. The second clock signalfrequency may be a multiple of the first clock signal frequency.

The input digital data streams may comprise at least two audio datastreams and/or at least one stream of control data. The control data maybe data for controlling the operation of a receiver of the least oneaudio data stream.

The first clock frequency may be equal to the sample rate of the atleast one digital data stream. The input data may be oversample data andthe sample rate of the at least one digital data stream is a multiple ofa standard audio frequency sample rate, such as 48 kHz or 44.1 KHz.

The pulse generator may be configured such that there is at least oneperiod defined by the second clock signal at the start of the transferperiod before the start of a data pulse and/or at least one perioddefined by the second clock signal at the end of a data pulse before theend of the transfer period.

At least one input data stream encoded by the data pulses may comprise abi-phase encoded data stream, for example a Manchester differentialencoding data stream. The transmission circuitry may therefore comprisea bi-phase encoder for receiving a first input data stream and producingthe bi-phase encoded input data stream. The bi-phase encoder may beconfigured to generate a bi-phase encoded data stream comprising atleast one illegal bit sequence that is not used for encoding data. Theillegal bit sequence may be inserted into the bi-phase encoded datastream to define a frame of data. The illegal bit sequence may comprisethree instances of the same value in the bi-phase encoded input datastream and in some embodiments the bi-phase encoder may generate twoinstance of the same data value into the bi-phase encoded data stream atthe start of a frame with the data value matching the last data value ofthe previous frame.

The digital data transmission circuitry may additionally comprise alogic unit for performing a logical XOR operation between the currentvalue of an input audio data stream and the current value of thebi-phase encoded data stream and the pulse modulator is responsive tothe bi-phase encoded data stream and the output of the logic unit.

In some embodiments at least some data pulses having the same durationencode different data values.

The pulse generator may be configured such that the rising or fallingedge at the start of the data pulse occurs substantially no later thanhalf way through the transfer period and/or the rising or falling edgeat the end of the data pulse occurs substantially no earlier than halfway through the transfer period.

The pulse generator may configured such that the set of possible datapulses that may be transmitted comprises at least one data pulse whichhas a duration which is unique within the set. The set of possible datapulses may comprise a minimum duration data pulse which has a durationwhich is unique within the set and which is shorter than the duration ofany other data pulse within the set. The circuitry may be operable in asynchronisation phase to transmit a sequence of data pulses comprisingat least one data pulse which has a duration which is unique within theset.

The pulse generator may be configured such that at least one possiblecombination of input data can be encoded by more than one possible datapulses. Over time, different instances of said combination of input datamay be encoded by different ones of said possible data pulses. Thedifferent possible data pulses for encoding a given combination of inputdata may have different durations and the pulse may select between thepossible data pulses so as to minimise any DC imbalance over time in thetransmitted data pulses. One of the possible data pulses for encoding agiven combination of input data may have a duration of greater than halfthe transfer period and another of the possible pulses may have aduration of less than half the transfer period.

The pulse generator may be configured such that there is at least onepossible data pulse that is not used to encode the input data but whichmay be transmitted by the pulse generator for synchronisation and/orcontrol.

At least one of the data inputs may be configured to receive an inputdigital data stream which is a n-bit data stream. In some embodimentsn=1. At least one input digital data stream may be a PDM data streamand/or at least one of said data inputs may be configured to receive aninput digital data stream which is a PCM data stream.

The digital data transmission circuitry as claimed may comprise acombination module for receiving at least first and second data streamsand producing a combined data stream as one of said input digital datastreams, wherein said combined data stream comprises one or more bits ofthe first audio data interleaved with one or more bits of the seconddata stream.

The digital data transmission circuitry may have a first data outputterminal associated with transmission of said data pulses to a receiverand said first output terminal is the only output terminal associatedwith said data transfer. Alternatively there may be a first data outputterminal associated with transmission of said data pulses to a receiverand a second output terminal for transmitting a clock signal fordefining the transfer period.

The output from the pulse generator may be connected to an audio signalpath on a printed circuit board of a host device and/or a connector of ahost device, which may for example be a socket. The connector maycomprise a connection for an audio accessory. In some embodiments thecircuitry is configured, in use, to provide audio data and power to aperipheral connected to said connector. The connector may haveconnections for audio data-out, power and ground. In some instances thepower connection also serves as said audio data-out connection. Theremay additionally be an audio data-in connection. The audio data-inconnection may also serves as the audio data-out connection. Theconnector may be an optical connector.

The digital data transmission circuitry may comprise bi-directionalinterface circuitry configured to transmit said data pulses generated bythe pulse generator over a first communications link and receive datapulses via said first communications link. The pulse generator may beconfigured to transmit data pulses during a first transfer period andthe bi-directional interface circuitry is configured to receive datapulse during a second, different transfer period. The bi-directionalinterface circuitry may comprise a drive circuit for voltage modulatingthe first communications link based on the data pulses and a readcircuit responsive to the resultant voltage on the first communicationslink, wherein the read circuit is configured to subtract the drivevoltage modulation from the resultant voltage signal.

Embodiments of the invention also relate to a digital receiver. Thus inanother aspect of the invention there is provided digital data receivercircuitry comprising:

-   -   an input for receiving a series of data pulses,    -   a sampler for sampling each received pulse within a transfer        period defined by a first clock signal such that there is a        single data pulse with a rising edge and a falling edge in each        transfer period, said sampler being configured to provide an        indication of which of a set of possible data symbols the data        pulse corresponds to based on the timing of the occurrence of        both the rising and falling edges of the data pulse within the        transfer period; and    -   decoding circuitry for generating at least one received digital        data stream based on said indication,    -   wherein said decoding circuitry is configured such that a        plurality of possible data symbols may be decoded as the same        value of a data bit of a received digital data stream.

The decoding circuitry may generate at least two received digital datastreams based on said indication.

The digital data receiver circuitry may comprise a clock recoverycircuit for recovering a clock signal from said series of data pulsesand generating said first clock signal. The clock recovery circuit maygenerate a second clock signal at a frequency which is a predefinedmultiple of the frequency of the first clock signal. The second clocksignal may be generated so as to be synchronised to the timing of therising and falling edges of the data pulses within the transfer period.

The set of possible data symbols that may be received may comprise atleast one data symbol which has a pulse duration which is unique withinthe set. In which case the digital data receiver circuitry may beconfigured to identify receipt of said at least one data symbol whichhas a pulse duration which is unique within the set and use receipt ofsuch a data symbol to synchronise the start and end of the transferperiod. The receiver circuitry may be operable in a synchronisationphase to receive a sequence of data pulses comprising at least one datasymbol which has a pulse duration which is unique within the set and toderive said first and second clock signals, with said first clock signalbeing synchronised to the transfer period.

In some embodiments however the digital data receiver may have a clockinput for receiving a first clock signal defining the transfer period.In which case there may be circuitry for generating a second clocksignal from said received first clock signal, the frequency of thesecond clock signal being a predefined multiple of the frequency of thefirst clock signal, wherein the second clock signal is generatedsynchronised to the timing of the rising and falling edges of the datapulses within the transfer period.

The received digital data streams may comprise at least two audio datastreams and/or at least one stream of control data. The control data maybe data for controlling the operation of an audio component whichcomprises said digital data receiver circuitry.

At least one received data stream encoded by said data symbols may be abi-phase encoded data stream, for example a Manchester differentialencoded data stream. A bi-phase decoder may decode said bi-phase encodeddata stream. The bi-phase decoder may be configured to identify at leastone illegal bit sequence that is not used for encoding data. The illegalbit sequence may be identified to define a frame of data. The illegalbit sequence may comprise three instances of the same value in thebi-phase encoded data stream. On detection of three instances of thesame value in the bi-phase encoded data stream, the data pulses encodingthe second and third instances may be identified as the first two datapulses of a new frame.

The receiver circuitry may also comprise a logic unit for performing alogical de-XOR operation between the current value of a first receiveddigital data stream and the current value of the bi-phase encoded datastream and the output of the logic unit is used as the output data forthe first digital data stream.

The data recovery circuit may output at least one received digital datastream which is a 1-bit PDM data stream and/or at least one receiveddigital data stream which is a PCM data stream.

The receiver circuitry may have a separation module having an input fora received data stream and configured to produce at least first andsecond data streams, wherein said first and second data streams arerecovered by taking one or more bits from the received data stream thatcorrespond to the first data stream and which are interleaved with oneor more bits of the second data stream.

In some embodiments there may be at least a first data extraction moduleand a second data extraction module wherein the first data extractionmodule and the second data extractions module are configured todetermine data values for different received data streams to oneanother. The first data extraction module may be configured to receivethe input data pulses from the interface and to pass said data pulses tothe second data extraction module. Each of the at least first and seconddata extraction modules may be associated with respective first andsecond audio transducers.

In some embodiments the circuitry may comprise power circuitryconfigured to derive a power supply from said input serial pulse-lengthmodulated audio data signal. The sampler and decoder circuitry may bepowered by said power supply.

The digital data receiver circuitry may be implemented in an accessorydevice such as a headset.

The invention also relates to a digital data transceiver circuitrycomprising digital data transmission circuitry as described above tosend digital data and digital data receiver circuitry as described aboveto receive digital data.

In a further aspect of the invention there is provided a digitalinterface comprising:

-   -   a data encoder for receiving at least two streams of digital        data to be transmitted and a first clock signal, and generating        a series of data pulses at a digital data output, said series of        pulses being encoded according to a first digital data protocol        such that there is a single data pulse having a rising and        falling edge in each of a plurality of transfer periods defined        by said first clock signal, the at least two streams of digital        data to be transmitted being encoded by the timing of the        occurrence of the rising and falling edges of the data pulse        within the transfer period; and    -   a data decoder for receiving a series of encoded data pulses at        a digital data input and decoding at least one received data        stream, said encoded data pulses being encoded using said first        digital data protocol at a data input.

The digital interface may be configured as a master circuit forcontrolling a digital data bus from said master to at least one slavecomponent. In which case the data encoder may be configured to transmitdata to a plurality of slave components connected in a chainconfiguration and said data decoder is configured to received data fromthe last slave component in the chain. The interface may be configuredto transmit audio data for at least one slave component having an audiooutput transducer and control data for controlling at least one slavecomponent. The control data may be transmitted in frames.

Alternatively the digital interface may be configured as a slave circuiton a digital data bus.

In some embodiments the digital decoder may be configured to receivedata pulses at said digital data input from an upstream audio circuit,decode said data pulses, identify any audio data and/or control data forsaid audio component, and the digital encoder may be configured togenerate data pulses for a downstream audio component based on saiddecoded data and any required modifications of the control data. Thecomponent having the interface may comprise at least one source of audiodata and the data pulses generated for the downstream component mayencode audio data from said source of audio data.

The invention also relates to a system comprising at least first andsecond components having a signal path between them such that data to betransmitted to or received from the second component is transmitted viathe first component, wherein the first component comprises a digitalinterface as described above. The first component may be configured toreceive a series of said data pulses encoding data for at least onecomponent and to forward a signal to said second component. The seriesof data pulses may encode data for said first component and the firstcomponent may be configured to extract said data for said firstcomponent. The first component may also forward the series of datapulses received to the second component. The first component may decodethe data in the series of data pulses received and use at least some ofsaid decoded data to generate a series of data pulses for transmissionto the second component. The first component may generate the series ofdata pulses for transmission to the second component to encode datagenerated by said first component. The system may comprise a pluralityof audio components with each component being connected to the nextcomponent in a chain.

Any of the circuits discussed above may be implemented as an integratedcircuit and/or may be embodied in an electronic device, which may be atleast one of: a portable device; a battery powered device; acommunication device; a computing device; a personal media player; amusic player; a mobile telephone; a docking station for a portabledevice; a headset; and a hearing aid.

The invention also relates to methods of data transfer. Thus in anotheraspect of the invention there is provided a method of digital datatransfer comprising defining a transfer period based on a clock signal,transmitting a single data pulse within each transfer period such thateach data pulse a distinct rising and falling edge within the transferperiod and encoding data at least two input digital data streams bysetting times of occurrence of the rising and falling edges of the datapulse within the transfer period.

The method may be performed using any of the implementations discussedabove in respect to the other aspects of the invention.

Embodiments of the invention allow low latency data transfer which isparticularly appropriate for transfer of digital data from one audioapparatus to another, for example for noise cancellation. In anotheraspect of the invention there is provided an audio apparatus comprising:

-   -   at least first and second audio input transducers; and    -   a connector for connecting said audio apparatus to an electronic        device, said connector having at least three contacts,        configured such that:    -   a first of said contacts provides a supply current for said        apparatus;    -   a second of said contacts is a ground return; and    -   a third of said contact carries a first digital composite signal        comprising oversampled signals from said at least two audio        input transducers.

The signals from said at least two audio input transducers mayoversampled at a data rate of at least 700 kilosamples per second or atleast 3 megasamples per second.

The apparatus may comprise a data encoder responsive to said audio inputtransducers to generate said first digital composite signal. The dataencoder may comprise a data transmitter according to the first aspect ofthe invention. The data encoder may be configured to generate said firstdigital composite signal by generating one data pulse within each of aplurality of transfer periods defined by a clock signal, each pulsehaving a single rising edge and a single falling edge; wherein the timeof one or both of the rising edge and the falling edge of the data pulseencodes the current respective samples of both the saiddelta-sigma-modulated audio input data signals.

The audio input transducers may be microphones. The apparatus may be anaudio accessory apparatus for use with an electronic device, for examplethe audio accessory apparatus may be a headset.

The connector may comprise a male-type plug, for example a 3.5 mm TRS orTRRS jack plug.

The at least two audio input transducers may be configured to detectambient noise. The apparatus may comprise at least two loudspeakers foraudio playback wherein said at least two audio input transducers areconfigured to detect ambient noise in the vicinity of said loudspeakers.The connector may therefore comprise at least a fourth contact forreceipt of audio data for said loudspeakers. The apparatus may comprisereceiver circuitry for recovering first and second audio signals from asecond composite digital audio signal received via said fourth contact,and conversion circuitry for applying the first and second audio signalsto said speakers. The receiver circuitry may be receiver circuitryaccording to the aspect of the invention discussed above. In particularthe second digital composite signal may comprise one data pulse withineach of a plurality of transfer periods, each pulse having a singlerising edge and a single falling edge, wherein the time of both therising edge and the falling edge of the data pulse encodes the currentsamples of both the said delta-sigma-modulated speaker signals.

Embodiments of the invention also relate to an electronic devicecomprising a device connector for connecting to the apparatus connectorof the audio apparatus according to this aspect of the invention, theelectronic device comprising a decoder for receiving said first digitalcomposite signal, to extract separate audio input signals, to calculaterespective signals to cancel the ambient noise, and to combine thesecorrection signals into the second composite digital signal as above fortransmission to the loudspeakers.

An audio ambient noise cancellation system comprising a device discussedabove connected to an apparatus as discussed above is also provided.

In another aspect of the invention there is provided audio interfacecircuitry comprising:

-   -   a pulse generator, responsive to at least one digital audio data        stream and at least one control data stream, to generate one        data pulse within each of a plurality of transfer periods        defined by a clock signal,    -   wherein each data pulse has a single rising edge and a single        falling edge and wherein the timing of at least one of the        rising edge and the falling edge of the data pulse within the        transfer period is based on a combination of the then current        values of said at least one digital audio data stream and said        control data stream.

The invention also provides an audio accessory comprising:

-   -   a plurality of wired connections to a connector, said connector        comprising:    -   a first connection for carrying supply current for circuitry in        the accessory;    -   a second connection for carrying ground return current for        circuitry in the accessory; and    -   a third connection for carrying a first oversampled digital        composite signal.

The connector may further comprise a fourth connection for carrying asecond oversampled digital composite signal.

In a yet further aspect the invention provides audio interface circuitrycomprising:

-   -   a pulse-length-modulator, responsive to a plurality of data        streams of audio data samples at a sample rate, to generate a        stream of data pulses at said sample rate;    -   wherein the length of each said data pulse is dependent upon a        combination of the then current audio data samples from said        plurality of data streams.

The audio data streams may comprise 1-bit digital audio data streamsand/or may correspond to audio data channels.

The pulse-length-modulator may be responsive to a first clock signalhaving a frequency equal to said sample rate. The pulse-length-modulatormay also be further responsive to a second clock signal, wherein thesecond clock signal has a frequency which is a multiple of the frequencyof the first clock signal and wherein the length of each data pulse isbased on a selected number of cycles of the second clock signal. Theminimum data pulse length may be a plurality of cycles of the secondclock signal. The maximum data pulse length may be shorter than theperiod of the first clock signal by a plurality of cycles of the secondclock signal.

The pulse-length-modulator may be configured so that at least onecombination of input audio data for said plurality of audio data streamscan be encoded as at least two different alternative data pulse lengths.The pulse-length-modulator may be configured so as to vary between saidat least two different alternative data pulse lengths when encoding saidat least one combination of input audio data and may alternate betweensaid at least two different alternative data pulse lengths when encodingsaid at least one combination of input audio data. Thepulse-length-modulator may randomly select one of said at least twodifferent alternative data pulse lengths when encoding said at least onecombination of input audio data and/or may, when encoding said at leastone combination of input audio data, vary between said at least twodifferent alternative data pulse length so as to control the averagedata pulse length for all instances of a given combination of inputaudio data. A plurality of combinations of input audio data may each beencoded as a plurality of alternative data pulse lengths and thepulse-length-modulator may vary between the respective alternative datapulse length so that the average data pulse length for each combinationof input audio data is substantially the same. A plurality ofcombinations of input data may each be encoded as two different lengthsof data pulse, the two lengths being symmetric about a predeterminedlength. One combination of input data may be encoded as a data pulsehaving a length substantially equal to said predetermined length.

The pulse-length-modulator may be configured to receive at least oneadditional data channel and wherein the length of at least someindividual data pulses encodes the then current audio data samples fromsaid plurality of data streams and also the then current data sample forsaid additional data channel. The additional data channel may be acontrol data channel. When no data is received on said additional datachannel the pulse length modulator may vary between pulse lengths thatencode the same audio data combination. When data is received on anadditional data channel the pulse-length-modulator may modulate a seriesof data pulses with a first reserved sequence prior to encoding saiddata of the additional data channel, where the first reserved sequencecorresponds to the encoding that would be used for a particular datasequence on the additional data channel and wherein thepulse-length-modulator is configured so as not to use the first reservedsequence when no data is available on the additional data channel. Whendata on the additional channel stops a second reserved sequence ofadditional data may be encoded.

The frequency of the second clock signal may be at least five times, orpossibly at least eight times, the frequency of the first clock signal.

The plurality of audio data streams may comprise left and right stereodata channels. The pulse-length-modulator may be configured to receive aseparate audio data stream for each of said audio channels. Thepulse-length-modulator may comprise a combiner for producing a combineddata value from each of said audio data streams.

A counter may be arranged to count at a frequency of the second clocksignal and a comparator may be arranged to compare the count value tothe combined data value.

The pulse-length-modulator may be configured to produce an output whichvaries between a first non zero voltage and a second non zero voltage.

The rising edges of the data pulses may be separated by a regular timeinterval equal to the sample period. The falling edges of the datapulses may be separated by a regular time interval equal to the sampleperiod.

The output from the pulse-length-modulator may be connected to an audiosignal path on a printed circuit board of a host device and/or to aconnector of a host device, such as a socket. The connector may comprisea connection for a headset. The pulse-length-modulator may beconfigured, in use, to provide audio data and power to a peripheralconnected to said connector. The connector may comprise connections foraudio data-out, power and ground. The power connection may also serve assaid audio data-out connection. There may also be an audio data-inconnection. In some embodiments the audio data-in connections alsoserves as the audio data-out connection. The connector may be an opticalconnector or an RF transmitter.

The audio interface circuitry may further comprising bi-directionalinterface circuitry configured to transmit said data pulses generated bythe pulse-length-modulator over a first communications link and receivepulse-length-modulated data pulses via said first communications link.The pulse-length-modulator may be configured to transmit data pulsesduring a first portion of the sample period and the bi-directionalinterface circuitry is configured to receive data pulse during a second,different portion of the sample period. The bi-directional interfacecircuitry may comprise a drive circuit for voltage modulating the firstcommunications link based on the data pulses and a read circuitresponsive to the resultant voltage on the first communications link,wherein the read circuit is configured to subtract the drive voltagemodulation from the resultant voltage signal.

In a further aspect of the invention there is provide an audio circuitrycomprising:

-   -   an interface configured to receive a serial pulse-length        modulated audio data input comprising a series of data pulses at        a sample rate;    -   data extraction circuitry configured to determine the pulse        length of said data pulse and to determine a data value for each        of a plurality of audio data streams from said pulse length.

The circuitry may comprise clock recovery circuitry configured torecover a first clock signal based on said sample rate. The dataextraction circuitry may be configured to sample the data pulse at apredetermined number of intervals within the period of the first clocksignal to determine the pulse length of the data pulse. The dataextraction circuitry may comprise a delay line having a plurality of tappoints. The data extraction circuitry may comprise at least a first dataextraction module and a second data extraction module wherein the firstdata extraction module and the second data extractions module areconfigured to determine data values for different audio data streams toone another. The first data extraction module may be configured toreceive the input serial pulse-length modulated audio data from theinterface and to pass said input serial pulse-length modulated audiodata to the second data extraction module. Each of the at least firstand second data extraction modules may be associated with respectivefirst and second audio transducers. The circuitry may further comprisepower circuitry configured to derive a power supply from said inputserial pulse-length modulated audio data signal. The data extractioncircuitry may powered by said power supply.

The audio circuitry may be implemented in a headset and/or may bearranged as a transceiver. An audio system is also provided.

In a further aspect there is provided a method of transmitting audiodata comprising, generating a pulse length modulated signal comprising aseries of data pulses at a sample rate;

wherein the length of each data pulse is dependent upon a combination ofthen current audio data samples from said plurality of data streams.

Aspects of the invention further relate to an audio system fortransferring audio data for a plurality of audio channels over a singlewire connection comprising a pulse-length-modulator configured toproduce a series of data pulses at regular intervals wherein the lengthof each data pulse encodes audio data for each of said plurality ofaudio channels.

Aspects of the invention further relate to data transfer system fortransferring data for a plurality of separate data channels over asingle wire connection comprising a pulse-length-modulator configured toproduce a series of data pulses at regular intervals wherein the lengthof each data pulse encodes data from each of said plurality of audiochannels.

Aspects of the invention further relate to audio interface circuitrycomprising:

a pulse-length-modulator responsive to audio data for a plurality ofaudio channels to generate data pulses at a regular time interval;

wherein the length of the data pulses encode said audio data; andwherein the length of an individual data pulse encodes audio data forsaid plurality of audio channels.

Aspects of the invention further relate to an audio interface forreceiving and encoding a plurality of streams of 1-bit audio datasamples and for transmitting a stream of encoded data pulses via asingle communication link.

Aspects of the invention further relate to an audio interface forreceiving and encoding a plurality of streams of 1-bit audio datasamples and simultaneously transmitting on a clock edge of a stream ofencoded data pulses a plurality of sampled 1-bit audio data samples viaa single communication link.

In a yet further aspect of the invention there is provided audiointerface circuitry for transfer of audio signals wherein said interfacecircuitry is operable in:

-   -   an analogue mode for receiving analogue audio signals; and    -   a digital mode for receiving digital audio signals        wherein, in said digital mode,    -   said digital audio signals comprise a series of data pulses        wherein the length of each pulse encodes at least two audio        digital data streams, and        wherein the interface circuitry decodes at least one of said        digital data streams.

The data may alternatively be encoded by the timing of the occurrence ofthe rising and falling edges of the data pulse within a transfer perioddefined by a first clock signal.

The interface circuitry may decode and condition at least one of saiddigital data streams.

The interface circuitry may comprise: at least a first analogue signalpath for receiving analogue audio data from a first contact on aconnector; at least a first digital path for receiving digital audiosignals from the first contact on the connector; and at least a firstswitch for enabling/disabling the first analogue signal path. It mayfurther comprise: a second analogue signal path for transferringanalogue audio data to/from a second contact on a connector; a seconddigital path; a second switch for enabling/disabling the second analoguesignal path. The at least first and second switches may default toenabling the first and second analogue signal paths and may comprisedepletion modes FETs within the first and second analogue signal pathsrespectively. The second analogue path may be for receiving an audiosignal from said contact and said second digital path may be fortransferring digital audio signals to said second contact. There mayalso be a third analogue signal path for transferring analogue audiodata to/from a third contact on a connector; a third digital path,wherein said third digital path is a path providing power to theinterface circuitry from a third contact of a connector; and a thirdswitch for enabling/disabling the third analogue path. The third switchmay default to enabling the third analogue signal path. The thirdanalogue signal may be for transferring analogue audio data to the thirdcontact of the connector.

The interface circuitry may comprise circuitry configured to, in thedigital mode, derive power from digital audio signals received on saidfirst digital data path.

The first and second analogue signal paths may be for receiving audiodata for respective loudspeakers. The third analogue signal path may befor transferring audio data to the connector from a microphone. Digitaldecoding circuitry may be coupled to said first digital path configuredto decode said data pulses in the digital mode. Digital encodingcircuitry may be coupled to said second digital path configured toencode data pulses in the digital mode. Discrimination circuitry may beconfigured to, when power is available on the third digital path, todetermine whether to operate in the digital mode and, if so, generate acontrol signal to disable the analogue signal paths. The discriminationcircuitry may be configured to, when power is available on the thirddigital path, to attempt handshaking with a device connected via saidconnector.

There may be a ground path for connecting to a ground contact on aconnector in both said digital and said analogue mode.

The audio interface circuitry may be implemented as an integratedcircuit. An audio apparatus may include such audio interface circuitryand at least a first audio output transducer, where said interfacecircuitry is configured to forward a received analogue audio signal tosaid first audio output transducer in the analogue mode and to decodesaid digital audio signal and forward a decoded digital data stream tosaid first audio output transducer in the digital mode.

The audio apparatus may comprise at least first and second audio outputtransducers wherein, when enabled, said first and second analogue signalpaths provides a direct connection to said first and second audio outputtransducers respectively. Digital decoding circuitry may comprise firstand second analogue outputs for connecting to said first and secondaudio output transducers respectively. The first and second switches mayswitch between the first and second analogue signal paths and the firstand second outputs of the digital decoding circuitry. The first andsecond signal paths may be connected to first and second converters forconverting between analogue and digital signals such that any analogueaudio signals received for the transducers are converted tocorresponding digital signals. Digital processing circuitry may beconfigured to apply digital processing to the digital signals outputfrom said first and second converters. The digital processing circuitrymay comprise analogue outputs for connecting to said first and secondtransducers respectively.

The interface circuitry may be connected to a connector and saidconnector is a TRRS jack. The apparatus may be an accessory apparatusconfigured to be connected to an electronic device, such as an audioheadset.

Aspects of the invention also provide a method of receiving audiosignals comprising: receiving said audio signals at an interfaceoperable in analogue mode and also in digital mode and selecting a modeto operate in, wherein:

in said analogue mode at least one analogue audio signal is received andforwarded for an audio output transducer; and

in said digital mode at least one digital audio signal is received, saiddigital audio signal comprising a series of data pulses wherein thelength of each pulse encodes at least two audio data streams, and theinterface circuitry decodes at least one of said data streams andforwards said digital data stream for an audio output transducer.

The method may comprising determining whether a connected devicetransmitting the audio signals is capable of sending signals suitablefor the digital mode and, if so operating in the digital mode, otherwiseoperating in the analogue mode.

Aspects of the invention also provide interface circuitry for receivingaudio signals wherein said interface circuitry is operable:

in an analogue mode to receive analogue audio signals; and

in a digital mode to receive digital audio signals wherein, in saiddigital mode, the audio data comprises a series of data pulses at apulse rate wherein the length of each pulse encodes at least two audiodata streams.

In a further aspect there is provided interface circuitry fortransmitting audio signals to an apparatus external to a host devicewherein said interface circuitry is operable:

in an analogue mode to transmit analogue audio signals; and

in a digital mode to transmit digital audio signals wherein, in saiddigital mode, the audio data comprises a series of data pulses at apulse rate wherein the length of each pulse encodes at least two audiodata streams.

In a further aspect there is provided audio interface circuitrycomprising:

at least first and second analogue signal paths for transmittinganalogue audio data to respective first and second contacts on aconnector;

at least a first digital path for transmitting digital audio signals tothe first contact on the connector; and

at least a second digital path coupled to a second contact on theconnector.

The second digital path may be for receiving digital data from thesecond contact in digital mode. A third analogue signal path may beprovided for receiving analogue audio data from a third contact on aconnector; with a third digital path for transmitting power to the thirdcontact on the connector in digital mode. A first interface may beprovided for digital communication with a first component of a hostdevice and a second interface for digital communication with a secondcomponent of a host device.

There may be a digital only path for communication between said firstand second interfaces. The first component may comprise an applicationsprocessor, a baseband processor, a transmission codec or a wirelesscodec.

The invention also provides, in a further aspect, a headset forreceiving audio signals comprising at least one audio output transducerand a connector for connecting to an audio device wherein the headset isoperable in

a digital mode for receiving digital audio signals via said connectorand applying said digital audio signals to said at least one audiooutput transducer and

an analogue mode for receiving analogue audio signals via said connectorand applying said received analogue audio signals said at least oneaudio output transducer.

In a yet further aspect there is provided an audio apparatus comprising:

at least first and second audio output transducers;

a connector for connecting the audio apparatus to another device havingat least first and second contacts;

interface circuitry operable:

in a first mode to enable first and second analogue signal paths fromthe first and second contacts directly to the first and second audiooutput transducers respectively; and

in a second mode to disable said first and second analogue signal pathsand provide at least a first digital signal path from the first contactto a digital decoder.

In said digital mode the interface circuitry provides power to thedigital decoder from the second contact.

BRIEF DESCRIPTION OF THE DRAWINGS

The invention will now be described by way of example only withreference to the following drawings, of which:

FIG. 1 illustrates a prior art three-wire data transfer protocol forstereo audio data;

FIG. 2 illustrates a data transfer system according to an embodiment ofthe present invention and FIG. 2b illustrates another embodiment of adata transfer system;

FIG. 3 illustrates examples of symbols that may be transmitted;

FIG. 4 illustrates one embodiment of suitable transmitter circuitry;

FIG. 5 illustrates the data waveforms on various signals lines of theembodiment shown in FIG. 4;

FIG. 6 illustrates one embodiment of suitable receiver circuitry;

FIG. 7 illustrates an alternative arrangement for providingsynchronisation in the receiver circuitry shown in FIG. 6;

FIG. 8 illustrates a set of symbols used in one embodiment;

FIGS. 9a and 9b illustrate embodiments of the invention arranged as achain of audio components;

FIGS. 10a and 10b show examples of slave device configurations;

FIG. 11 illustrates example data waveforms on the various signal linesof the embodiment illustrated in FIG. 2b when using a fixed rising edge;

FIG. 12 illustrates one embodiment of a suitable audio transmitter forthe embodiment shown in FIG. 11;

FIG. 13 illustrates example waveforms on the various signal lines of thetransmitter embodiment illustrated in FIG. 12;

FIG. 14 illustrates one embodiment of a suitable audio data receiver forthe embodiment shown in FIG. 11;

FIG. 15 illustrates example waveforms on the various signal lines of thereceiver embodiment illustrated in FIG. 14;

FIG. 16 illustrates one example of an encoding scheme which allows datainput combinations to have alternative possible encodings;

FIG. 17 illustrates a data transfer system according to anotherembodiment of the present invention using a fixed rising or fallingedge;

FIG. 18 illustrates another example of an encoding scheme;

FIG. 19 illustrates an embodiment of interface circuitry for sending PLMdata in both directions over a single data link;

FIG. 20 illustrates example waveforms on the various signal linesillustrated in FIG. 19;

FIG. 21 illustrates another embodiment of interface circuitry forsending PLM data in both directions over a single data link;

FIG. 22 illustrates an application of an embodiment of the invention;

FIG. 23 illustrates an application of another embodiment of theinvention;

FIG. 24 illustrates an application of a further embodiment of theinvention;

FIG. 25 illustrates an application of another embodiment of theinvention;

FIG. 26 illustrates an application of a yet further embodiment of theinvention;

FIG. 27 illustrates an application of a further embodiment of theinvention;

FIG. 28 illustrates an application of another embodiment of theinvention;

FIG. 29 illustrates various consumers and generators of data in a deviceand a connected accessory apparatus;

FIG. 30 illustrates a first embodiment of the invention for transferringdata in the arrangement shown in FIG. 29;

FIG. 31 illustrates a second embodiment of the invention fortransferring data in the arrangement shown in FIG. 29;

FIG. 32 illustrates a connection interface for an accessory according toan embodiment of the invention;

FIG. 33 illustrates a connection interface for an accessory according toanother embodiment of the invention;

FIG. 34 illustrates a connection interface for a device according to anembodiment of the invention;

FIG. 35 illustrates an electronic device with various connections foraccessory apparatuses; and

FIG. 36 illustrates various connections between internal and externalsystems of an electronic device.

DETAILED DESCRIPTION

As mentioned above increasingly audio data is being stored andtransmitted in digital form. For instance in a portable device such as amobile telephone or tablet computing device, there may be an audiodigital signal processor or audio codec which processes the audio dataand which transmits audio data to and from audio transducers suchloudspeakers and/or microphones which may be provided on the host deviceand/or which may be connected to the host device via a suitableinterface. There are advantages in transmitting such data to and fromthe transducers in a digital format but conventional digital transferoften requires multiple signal paths, e.g. for a data signal, a bitclock and a frame clock as described previously with reference toFIG. 1. Providing multiple wires, or conductive tracks, routed aroundthe device all carrying various signals can be disadvantageous in termsof cost, electromagnetic interference and power consumption. There istherefore a desire to transmit data, especially audio data, betweencomponents using minimal wires carrying maximal data. There is also adesire for low latency data transfer, especially in applications such asambient noise cancellation.

Embodiments of the present invention relate to data protocols, datainterfaces and methods and apparatus for data transfer that allow highspeed/low latency and accurate digital data transfer of multiplechannels of data that can be implemented with very few wires/links—andin many cases a single link. The embodiments have a simple physicalinterface with fewer connections than many other known data interfacesoperating other data protocols. The embodiments are suitable forpoint-to-point connection of audio producing and consuming componentsand also connecting several components of a device, thus providing ahigh speed audio data bus without requiring several parallel links.Embodiments also relate to connection of peripheral devices, oftenreferred to as “accessories”, such as headsets or the like which can usedigital data transfer and which may also be compatible with legacysystems and connection components. The low latency of the data transferis very suitable for applications such as ambient noise cancellation.

FIG. 2 illustrates one embodiment of a data transfer system fortransferring digital data over a link 203, i.e. conductive path, betweena first component 201 and a second component 202. The apparatus issuitable for transferring multiple simultaneous channels of data over asingle link. In this example stereo audio data is transferred,optionally along with control data for controlling the set-up andoperation of transducers such as microphones and speakers for example.The first component 201 may be audio interface circuitry that isincorporated as part of an audio codec or a digital signal processor ofa host device for example. The second component 202 may be audiointerface circuitry, for example as part of transducer drivingcircuitry. The second component 202 may be part of the same host deviceas the first component 201 and thus the link 203 may comprise a trace ona PCB of the host device or other permanent wired connection.Alternatively the second component 202 could be part of a separatedevice to the first component 201. For example the first component 201could be located in a media player or mobile telephone or the like andthe second component 202 could be part of a second device, for example aperipheral or accessory device, that is coupled in use such as a headsetor docking station or the like. In this case the link 203 may comprise aconductive path involving a suitable connection (not shown) such as aplug and socket. In further embodiments the first and second componentsmay both be implemented on the same integrated circuit, and the link isan on-chip connection, to reduce the width of interconnect bussingbetween circuit blocks. Various applications of the embodiments of thepresent invention will be described in more detail later.

The audio interface circuitry of the first component 201 comprises amodulator which receives audio data input data, PDM-R and PDM-L, for theleft (L) and right (R) audio channels. In this example the L/R inputdata are separate 1-bit, i.e. Pulse Density Modulation (PDM) digitalinput data streams for each channel, i.e. left channel and rightchannel. In some applications the modulator may also receive controldata CTRL to be transmitted to the second component for controllingtransducers for example.

The modulator implements a protocol to transmit data that ensures thatdata, possibly multiple channels of data, and a clock are encoded into asingle pulse stream.

Encoding

The encoding protocol operates by transmitting data in the form of datapulses in data transfer periods defined by a data transfer clock signal.The data pulses reversibly encode the data payload to be transmitted—inother words the original data payload can be decoded unambiguously fromthe data pulse. The transfer period is defined by a data transfer clocksignal which may be derived from a supplied clock signal, for instanceeach data transfer period may correspond to a single cycle of a suppliedclock signal, though in some cases the transfer period may be derived asa multiple or as a fraction of the period of a supplied clock. Theencoding protocol transmits one pulse in each transfer period such thateach data pulse has one distinct rising edge and one distinct fallingedge. In other words the transmit, i.e. encoding, protocol ensures thatthere is no more than one data pulse per transfer period and that thereis always a gap between successive data pulses. This ensures that withineach transfer period there will be a single rising edge and a singlefalling edge associated with each data pulse.

With these requirements in place the data is encoded by modulating thetiming of the rising and falling edges of the data pulse within thetransfer period. In other words the occurrences of the rising andfalling edges of the data pulse are quantised in time, i.e. they occurat one of a fixed number of possible times in the transfer period. Thequantisation of the rising and falling edges in time thus ensurereversible encoding (i.e. ensures that the original data payload can bedecoded by looking at the times at which the rising and falling edgesoccur within the transfer period). In effect the location and durationof each data pulse within the relevant transfer period defines a symbolwhich is used to encode the data. Advantageously, ensuring only a singlerising edge and single falling edge can occur per transfer period helpsto minimise both power consumption and EMI issues.

Conveniently the transfer period is defined by a first clock signal TCLKand the rising and falling edges of the data pulse are defined by, i.e.synchronised to, a second clock signal MCLK, the second clock signalfrequency being an integer multiple of the first clock signal frequencyf_(MCLK). For example consider that each transfer period is equal to oneperiod of the first clock signal TCLK and that the second clock signalMCLK is n times the frequency of the first clock signal TCLK. A transferperiod is therefore equal in length to n periods of the second clocksignal MCLK. The transfer period thus effectively comprises n time slotsdefined by the second clock signal MCLK. In one example the second clocksignal MCLK may have a frequency which is eight times the frequency ofthe first clock signal TCLK and thus the transfer period consists ofeight time slots defined by the second clock signal. A value of n whichcorresponds to a power of two is relatively easy to implement usingfrequency division/multiplication techniques although other values of ncould be implemented if desired. In the description below the firstclock signal will be referred to as a transfer clock signal TCLK or asymbol clock signal and the second clock signal will be referred to as amaster clock signal MCLK.

Within each transfer period a data pulse is generated by defining acertain number of contiguous time slots to be logic 1 and the remainingtime slots to be logic 0. Note that logic 1 and 0 are used herein tosimply indicate different states and do not imply any particular voltageor intensity levels etc. For instance in some applications using voltagelevels to transfer data, logic 1 may indicate high voltage and logic 0may indicate low voltage whereas in other applications logic 1 may below voltage and logic 0 may be high voltage. Likewise it wouldalternatively be possible to define a data pulse as a continuous periodof logic 0 with the rest of the transfer period being logic 1. For thepurposes of brevity and for explanation only however the data pulse willbe taken to be a continuous period of logic 1 and a transition fromlogic 0 to logic 1 will be referred to as a rising edge and a transitionfrom logic 1 to logic 0 as a falling edge.

Considering the need for each pulse to have a distinct rising edge and adistinct falling edge it will be appreciated that a period of logic 1 atthe end of one transfer period cannot be followed by a period of logic 1at the beginning of the next transfer period.

In one embodiment therefore the first time slot of each transfer periodmay be set to logic 0. Thus even if a transfer period has a pulse whichis logic 1 to the end of the transfer period there will be a distinctfalling edge at the end of the transfer period and a gap before therising edge of the pulse of the next time period. It will be appreciatedhowever that equally a period of logic 0 may be implemented at the endof each transfer period and/or there may be a period of logic 0 at thebeginning and end of each transfer period.

Taking a transfer period of eight time slots for example and ensuringthat the first time slot is always logic zero there are 28 differentcombinations of different timings of rising and falling edges for a datapulse, where each transfer period has a single data pulse and the risingand falling edges are synchronised to the time slots. More generally forn time slots, with the state of either the first or last time slotpredefined to ensure a gap between successive data pulses and assumingthat data pulses are allowed to differ by one time slot in duration, thenumber of possible symbols is equal to (n−1)+(n−2)+ . . . +1. As theskilled person will appreciate the sum of successive integers from 1 tom is equal to m·(m+1)/2 so the sum from 1 to (n−1) is equal to(n−1)·n/2. Thus with eight time slots there are 28 possible symbolswhich can be generated to provide data encoding using the exampleencoding protocol discussed above. These 28 possible symbols areillustrated in the left hand column of FIG. 3.

FIG. 3 illustrates the eight time slots of the transfer period and thevarious combinations of rising and falling edges—where rising andfalling edges are synchronised to the time slots, a minimum period ofone time slot between rising and falling edges is allowed and a fallingedge is allowed at the end of a transfer period.

According to embodiments of the present invention different symbols maytherefore be used to encode a different data value or combination ofdifferent data values. In some embodiments each possible symbol may beused to encode a data value, possibly with some data values beingencoded by more than one symbol, however in other embodiments only someof the different symbols that are possible may be used in practice.

With 28 possible symbols per transfer period a 4-bit digital data signalcan be encoded each transfer period. As a simple example consider howtwo 1-bit PDM audio data streams, for example stereo audio data PDM-Rand PDM-L, and a channel of audio related control data, CTRL, could betransferred. The audio data may be received at the modulator ofcomponent 201 at a first sample rate. The modulator may also receivecontrol data which may also be at the same first sample rate, althoughthe control data may be at a lower sample rate.

It will be appreciated that audio data is typically stored digitally atone of a few standard sample rates, for instance 48 kilosamples persecond. Such audio data may be converted and/or interpolated into ahigher sample rate PDM audio stream for transmission and thus the samplerate of the PDM audio for transmission may be higher than the samplerate of the base audio. Thus if the underlying audio sample frequency,f_(S), is say 48 kHz, the PDM data sample rate may be say 64 f_(S) orapproximately 3 MHz. The data stream transmitted is therefore anoversample data stream, i.e. a data stream at a data rate which isgreater than the underlying sample rate of the data being transferred.The first clock signal frequency, i.e. the transfer clock signalfrequency f_(TCLK) may be the same frequency as the PDM input datastreams. With eight time slots per transfer period for example themaster clock frequency MCLK, i.e. that of the second clock signal, wouldthen be approximately 24 MHz. With four bits of data coded per symbolthe data rate could be about 12 Mb per second.

In one embodiment the modulator may therefore operate with a transferperiod equal to the sample rate period of the incoming data, although inother embodiments the transfer clock signal TCLK defining the transferperiod may be a multiple of the sample rate of the incoming datastreams. Each transfer period the modulator may generate a data pulsewithin the transfer period whose rising and falling edges, i.e. thelocation of such edges within the transfer period, are defined by thestate of input data. Table 1 below shows an example of how the inputdata may be encoded.

TABLE 1 RE Time FE Time PDM-L PDM-R CTRL Symbol End of slot End of slot0 0 0 1 1 8 0 0 1 2 1 6 0 1 0 3 1 4 0 1 1 4 2 8 1 0 0 5 2 6 1 0 1 6 2 41 1 0 7 3 8 1 1 1 8 3 6

The columns PDM-L, PDM-R and CTRL illustrates the various datacombinations that may be input on left and right audio data streams anda control data stream. The symbol column illustrates a particular symbolnumber (which is just a label to distinguish between different possiblesymbols). The RE time and FE time columns illustrate the end of the timeslot at which the rising edge and falling edge occur for that particularsymbol. Note that the symbol allocation to data combinations in thisexample is simply to explain the principles: the actual correspondenceof data to symbol may vary considerably.

It can be seen in this example that earliest rising edge of any symbolis at the end of time slot 1, thus ensuring at least one time slot oflogic 0 at the start of a transfer period. In this example a fallingedge can therefore occur at the end of time slot 8. It will beappreciated that symbol 2 has a data pulse duration of five time slots,as does symbol 7. Thus the data pulses of different symbols may be ofequal duration but the symbols may be distinguished by the differentrising edge and falling edges times within the transfer period. It willalso be noted that the time of the rising edge and the falling edges ofthe data pulses vary from symbol to the symbol and are quantised to thetime slots. A transmitter may therefore receive the input data streams,identify the current combination of input data and the correspondingsymbol and generate a data pulse that corresponds to that symbol. Areceiver may therefore receive the data pulse, identify the timing ofthe occurrence of the rising and falling edges within the transferperiod and thus identify the relevant symbol and thus identify thecombination of input data.

As used in this specification the term PLM (pulse length modulation)will be used to refer to this type of data encoding. It will, of course,be clear from the foregoing however that two symbols which encodedifferent combinations of input data may have the same pulse durationand thus data is not encoded solely by pulse length (but also by pulseposition within the transfer period) and thus the term PLM as used inthis specification should be read accordingly.

Data Transmission

Component 201 of FIG. 2 is thus arranged to transmit data pulses encodedby the use of different symbols as described above. Component 201therefore comprises a modulator for producing an appropriate symbol forthe combined input data. The modulator receives the incoming datastreams to be transmitted and may receive a master clock signal MCLKwhich can be used to determine the time slots for the symbols. Themodulator may also receive a transfer clock signal TCLK (or may generateTCLK based on the master clock signal MCLK) which defines the transferperiods, i.e. which sets the symbol rate of the communication link. Eachperiod of the transfer clock signal TCLK the modulator may determine theappropriate symbol to encode the then-current input data, for instanceby consulting an appropriate look-up-table. Based on the requiredsymbol, i.e. which symbol is identified with that data combination inthe look-up-table, the modulator will then generate a data pulse withappropriate rising and falling edges, the timing of the rising andfalling edges being determined by the master clock signal, MCLK. Themodulator therefore acts as a pulse length modulation (PLM) modulator.It will of course be appreciated that the modulator may instead receivethe transfer clock signal TCLK and generate the master clock signalMCLK, e.g. by some frequency multiplier or phase-lock loop circuitry, orreceive some other clock signal and thence generate the transfer andmaster clock signals.

Note whilst the example above has discussed transmitting a data pulseencoding multiple 1 bit input data streams the same principles apply tomulti-bit streams. Thus, for example, two 2-bit audio input streamscould be received and encoded using 16 possible symbols.

Equally, depending on the number of different possible symbols that canbe generated and the ratio of the transfer period to the sample periodof the input data, multiple bits of data from a given data stream couldbe encoded in a given symbol. Thus a given data rate can be achieved byencoding multiple samples of an input data stream in a pulse stream at areduced sample rate. In other words, two 1-bit input data streams couldbe encoded at a symbol rate at half the input data rate by encoding 2bits from each respective input data stream in each symbol.

In general each symbol encodes multiple bits of information. Asmentioned using a set of 16 different possible symbols allows 4-bits tobe encoded per symbol. Thus one symbol will encode say 1101 and adifferent symbol will encode 0101. In this specification the termchannel will be used to refer to data carried by a specific bit encodedby the symbol. Thus the data carried by the first bits of each symbolwill be seen as one channel, the data carried by the second bit of eachsymbol will be seen as a second channel and so on. How the data in thechannels corresponds to distinct data streams, i.e. data streamsproduced by or intended for a specific component such as an audiotransducer may be defined according to the application. Thus in theexample of transmitting stereo audio data the data stream for the rightloudspeaker may be received and encoded as a separate channel, i.e. thesymbols are selected so that the first bit say of the symbol payloadencodes the right audio data. The left audio data may then encode aseparate channel, thus the second bit of the symbol payload may encodethe left audio data. Control data may be encoded as the fourth bit ofeach symbol. (Note the terms first and second are used for convenienceonly—any given receiver may decode the data in a different order or onlydecode the data relevant to it depending on how the decoding isperformed, for instance a receiver for the right audio data may simplybe provided with a look-up-table that indicates a first sub-set of thesymbols indicates logic 1 for that receiver and a second sub-set ofsymbols indicates logic 0).

The data streams encoded by the pulse length modulation (PLM) modulatormay themselves comprise individual data streams for more than onecomponent. For instance two data streams could be merged into a singlestream by alternating the data in a predefined way. The merged streamcould be sent via one channel by the PLM modulator. A PLM receiver woulddecode the data from that channel to produce the merged data streamwhich could then be separated back into the individual data streams.

In some instances however an individual data stream could be encoded inmore than one channel of the PLM data.

Frames

In some applications the data may be transmitted in frames. A frame isdefined by a certain number of symbols. The use of frames can bebeneficial for the purposes of control data as certain bits in the framecan be reserved for particular functions. In addition the use of framescan allow multi-bit coding of the data channels, i.e. data words ofn-bits, where n>1, may be transmitted rather than just a 1-bit stream.Where the encoding protocol includes the use of frames there may be aframe synchronisation stage as will be described in more detail later.

The frames may comprise a set number of symbols, for instance 64symbols. Thus if each symbol encodes four bits, this allows fourchannels, each of 64 bits per frame. As will be described below howevera control channel may be bi-phase encoded so that there are actuallyonly 32 bits of control data per frame. Typically the control data canbe transmitted with a much lower data rate than the data channels so 64bits per frame may not be necessary and the use of bi-phase encoding canbe advantageous in terms of DC balance and frame synchronisation.

With bi-phase encoding each control bit comprises two ‘half-bits’ thatare encoded by successive symbols. One possible bi-phase encoding isManchester differential encoding. In Manchester Differential encodingone half bit is always logic 1 and the other half bit is logic 0, i.e.the two possible legal half-bit pairs are (1-0) and (0-1). Data isencoded by whether the first half bit of a half-bit pair maintains thesame state as the last half-bit of the preceding pair or changes state.For example if the first half-bit of a half-bit pair changes state fromthe previous half bit, then that half-bit pair (i.e. control bit) mayindicate logic 0. Thus the sequence (1-0) (1-0) where the last two bitsare a half-bit pair indicates that half-bit pair is logic 0. Likewise(0-1) (0-1) also indicates that the second half-bit pair is logic 0. Inthis example if the first half-bit of a half-bit pair maintains thevalue of the previous half-bit then the pair indicates logic 1, i.e. forboth (1-0) (0-1) or (0-1) (1-0) the second bit-pair encodes logic 1.

This bi-phase encoding scheme means that every legal half-bit paircontains individual half bits of logic 1 and logic 0. Thisadvantageously provides DC balance for the lines carrying the half-bitsand ensures that a long run of repetitions of the same symbol is notpossible on the PLM data line. It also means that for valid data no morethan two consecutive control half-bits can have the same data value.This means that a sequence of control data values having three instancesof logic 1 as the decoded half-bit values is an illegal state for datatransfer, i.e. the sequences 1-1-1 or 0-0-0 are not legal. Such anillegal state can however be used for frame synchronisation to indicatethe beginning of a frame.

For the control channel the use of three half bits of the same logicstate can thus be inserted at the beginning of every frame. Note thatthe first half bit of such a sequence could actually be the last halfbit of the previous frame. Thus the start of each frame could begin withtwo half-bits of the same state, the state being chosen to match thestate of the last half bit of the previous frame. The receiver willdetect three half bits of the same state and will then known that thelast two bits of such a sequence indicate the start of the frame. Thereceiver may maintain a count of received symbols and compare the actualcount between detection of the illegal state, i.e. frame synch, with theexpected number of symbols and generate an error signal if too many ortoo few symbols are detected.

By using Differential Manchester encoding as the bi-phase encodingscheme it is guaranteed that the last transition on the control datachannel will be half way through the last bit in a frame. Consequently,as mentioned the illegal state providing the Frame Synch of one framecan overlap the last half bit, of the previous frame. In other words thelast symbol in a frame can belong to both that frame, and the FrameSynch of the next frame. Unlike other bi-phase encoding schemesDifferential Manchester never causes a Frame Synch jitter of one symbolregardless of the value of the last bit in a frame. Other bi-phasecodings are known however, such as Manchester 1b2b and could be used insome implementations. In Manchester 1b2b encoding again each half bitpair always contains a 1 and a 0 with one of the two legal half-bitpairs indicating one state and the other indicating the other state.E.g. (0-1) may indicate logic 1 and (1-0) logic 0. This scheme alsoensures a near 50:50 ratio of 1s and 0s and again three successivehalf-bits of the same logic state represents an illegal state and can beused for frame synchronisation.

As mentioned the use of frames can provide dedicated control fieldswithin the frame. For example when used in a bus arrangement (as will bedescribed in more detail later) with multiple receivers a control fieldmay be used to indicate whether there is any valid data for a particularreceiver. The control frame may also comprise an address fieldindicating a particular receiver.

For example with a frame of 64 symbols with control data bi-phaseencoded there are 32 control bits per frame.

The first bit may, as described, be used for frame synchronisation andencodes an illegal bi-phase state. One control bit may be used as a BUSYbit which is asserted by the transmitter to indicate valid data is beingtransmitted. Some control bits may be used for addressing where there ismore than one receiver that receives the data, i.e. specifying a fixedaddress for a receiver or using an address nibble. In a chainedimplementation as will be described later at least control bit may bereserved for a slave device to communicate to a master device. There mayalso be one or more bits reserved to indicate a register address.

In some embodiments data registers of a receiver may be read or write orboth and so a transmitter can send or receive register data to areceiver by an appropriate register address selection using the controldata. Certain address locations may be reserved to store optional devicecapability and manufacturer identification information.

A control frame may therefore include an instruction to read data fromand/or write data to the control registers of a receiver. The receivermay decode and store the control data frame, determine whether theinstruction is for that receiver and, if so, action the relevantinstruction.

The frames can also be used for the data channels and allows data to beencoded using pulse code modulation (PCM) or similar techniques sendingdata words. Word synchronisation, i.e. identifying the most significantbits (MSBs) or least significant bits (LSBs), can be provided throughsynchronisation of the control frames.

Any number of any length words over any of the data channels may be sentup to a maximum of the frame length, i.e. the number of transfer clockperiods in each frame, e.g. 64 bits in the example described above.Selection of the PCM word length and the number of channels may becontrolled by a control register write to the appropriate device. Forexample one data channel could support 4 words of 16 bit PCM data usingall 64 bits in the complete frame.

It should be noted that the data channels encoded by the various symbolscan represent data which is encoded in different formats, i.e. at leastsome of the data channels may have different underlying encodings. Forinstance the various data channels could be PCM encoded with differentword lengths, or some data channels could correspond to PCM encoded dataand others to PDM encoded data. For example if three data channels of 64bits per frame are available one PLM data channel may be used to send aPDM data signal, e.g. a Dolby Digital™ bit stream (at a data rate equalto the transfer period data rate—which may for example be 64 f_(S) asdescribed above). The other two PLM data channels could be used to sendeight 16 bit PCM data streams via the other two PLM data channels.

As mentioned above different audio data streams may be sent over one PLMchannel by interleaving symbols for the various data streams. For PDMencoded data streams the symbols may typically be interleaved in analternating manner, i.e. two streams A and B may be transmitted bysending symbols encoding stream A alternating with symbols encodingstream B. For PCM encoded data the various words are typicallyalternated in the frames, i.e. a complete word for stream A may be sentfollowed by a complete word for stream B (although if desired thesymbols of the words could be interleaved). It would also be possible totransmit n symbols encoding PDM data followed by m symbols encoding aPCM data word provided that n and m were known, fitted into a frame andallowed synchronisation within the frame.

As described above where the control data is encoded using ManchesterDifferential encoding or a similar bi-phase encoding scheme every legalhalf-bit pair contains individual half bits of logic 1 and logic 0. Thusthe half-bits of the control data channel—which typically correspond toindividual bits of a data channel—have approximately equal numbers ofinstances of logic 1 and logic 0 over time. In some embodiments thecontrol channel half bits to be encoded by combined with the data streamfor one or more data channels in an XOR operation before the data streamis encoded. In other words the incoming data streams and control dataare received. The half-bit pair required to bi-phase encode the controldata is determined. Each data bit is then XORed with the relevantcontrol data bit before selecting an appropriate symbol. This ensuresthat any long run of 1s or 0s in the data stream are averaged toapproximately a 50:50 ratio of logic 1 to logic 0. The data is recoveredby decoding the symbol, identifying the value of the control half-bitand the bits for the data channels then a de-XOR operation is performedusing the value of the control half-bit.

Transmitter

FIG. 4 illustrates one embodiment of transmitter circuitry 400 forreceiving various input data streams of data bits, including a controldata stream, and transmitting a PLM data stream in frames. Thetransmitter circuitry includes a pulse generator, responsive to theinput digital data streams and to a first clock signal, to generate asingle data pulse having a rising edge and a falling edge within each ofa plurality of transfer periods defined by said first clock signal. Asdescribed above the time of occurrence of the rising edge and thefalling edge of the data pulse encodes the then current data bits ofsaid input digital data streams. FIG. 5 illustrates waveforms at variouspoints in this circuitry.

The transmitter circuitry receives (or generates) a frame clock FCLK, atransfer clock TCLK and master clock MCLK. As described above thetransfer clock TCLK defines the transfer period and the master clockMCLK defines the time slots within the transfer period. In this examplethere are eight time slots in the transfer period. The frame clockdefines the frame and in this example one period of the frame clock isequal to 64 periods of the transfer clock.

The transmitter also receives audio data and, in the example waveforms,shown it receives three streams of audio data: DATA1, DATA2 and DATA3.

In this example DATA1 is a single-bit waveform at a sample rate equal tothe frequency of the transfer clock, i.e. a sample rate of f_(TCLK).

DATA2 in this example comprises both left and right-channel data in aninterleaved 1-bit-PDM format, alternating between left and right, withthe data synchronised to the frame clock FCLK, so that left data appearsfirst in the frame.

The third data stream DATA3 received comprises three data streams: astereo pair of 16-bit signals A[15:0] and B[15:0] and a single 24-bitsignal C[23:0]. These last three streams are each sampled at a samplerate of f_(FCLK) and may require conversion into a serial data streamDATA3S by the parallel-to-serial converter 401, or may be already in asuitable serial format. This third data stream DATA3 (or the convertedserial data stream DATA3S) in this example uses 56 (for the two 16-bitand one 24-bit data streams) out of the 64 transfer periods in theframe. The remaining 8 data bits in each frame may thus be zero for thisdata channel or may be set to some dummy data pattern.

The transmitter 400 also receives various control bits for onwardtransmission, up to 31 bits per frame. These are serialised to a samplerate of f_(TCLK)/2 if necessary by parallel to serial converter 402, andManchester encoded by bi-phase encoder 403 to provide a bi-phase encodedcontrol data stream CTRL_MCR. Note the Differential Manchester encodingillustrated gives a transition in CTRL_MCR every alternate TCLK cyclefor 31 out of the 32 such alternate clock cycles per frame. However theencoder forces no transition for the first transfer period in the frame.As noted above, this may be used by a receiver to synchronise theframe/control word and any multi-bit data.

As illustrated in FIG. 4, the data streams DATA1, DAT2, and DATA3S mayeach be input to a logic unit 404 to perform an XOR operation with thebi-phase control data CTRL_MCR. As mentioned above this can avoid longruns of zeros and/or push spectral energy away from low frequencies.

The modified data bits available in each transfer period are then usedto define the rising and/or falling edge of the output stream. FIG. 4illustrates the data bits being input into a look-up table 405, whichprovides a high or low level for each of the eight available cycles ofmaster clock MCLK, to give a data stream sampled at MCLK rate comprisinga single pulse within each transfer period, but with the timing of therising and falling edges in each transfer period defined according tothe entry in the look-up-table, or the like, corresponding to the set ofdata bits received.

Data Receiver

Referring back to FIG. 2, the receiver 202 receives the data pulses anddecodes, i.e. extracts the data by determining when the rising andfalling edges of the data pulse fall within the transfer period. Thiseffectively identifies the relevant symbol and hence the data encoded.

In order for the receiver 202 to recover the data it needs to be able toidentify the symbol that the data pulse corresponds to. The receiverthus typically determines the relevant time slots occupied by the PLMdata pulse in the transfer period, i.e. when the relevant rising andfalling edges lie.

In some embodiments the receiver establishes its own clock signals atthe master clock rate MCLK and transfer clock rate TCLK. In someinstances one clock signal, say the transfer clock signal TCLK, may berecovered from the modulated data signal and used to generate the otherclock signal. The nature of the transmit protocol, advantageouslyensures that there is a distinct data pulse with a rising and fallingedge each transfer period, and only one data pulse per transfer period,which enables a clock signal to be reliably recovered using the datastream itself without requiring a separate clock signal. As there isonly one pulse for each transfer period, a clock at the transfer clockrate TCLK and/or master clock rate MCLK can be established using knowntechniques, such as using phase locked loops and frequencydivision/multiplication. Having only one pulse for each transfer periodis a particular advantage of the embodiments of the present invention.In other words the clock and a plurality of data channels arerecoverably encoded for transmission over a single data link, i.e. wire,and the receiver can recover both a relevant clock signal and theencoded data from the recoverably encoded signal that is transmittedover the single link, thus advantageously avoiding the need to transmita clock or clocks over an otherwise additional parallel link.

For correct clock recovery the receiver may therefore comprise aphase-locked loop (PLL) or frequency-locked-loop (FLL) circuit togenerate a clock that has at least the speed of the master clock MCLK.

The protocol may involve an initial synchronisation period to recoverthe master clock signal, with the correct phase, from the received datapulses and also to ensure that the start/end of the transfer period isknown.

The master clock signal may be recovered using the PLL/FLL and the phaseadjusted so that the incoming data can be correctly strobed and thesymbol length and rise and fall times can be correctly measured.

The receiver may then synchronise the transfer period so that thesymbols can be correctly identified. This may involve determining thelength of the data pulses in terms of numbers of time slots and/or thetime between the end of one data pulse and start of the next andadjusting the possible start/end times of the transfer period to matchpermissible symbols and rule out any unused symbols. In essence once themaster clock signal is established the timing of the various time slotsis established. It is then necessary to identify the time slots whichcorrespond to the start and/or end of the transfer period. In theexample described above the end of the transfer period can coincide withthe end of a data pulse and must be at least one time slot before thestart of the next data pulse. The end of the transfer period may behypothesised to coincide with the end of any time slot that matchesthese criteria and the hypothesis updated if any illegal symbols aredetected—e.g. a data pulse running beyond the end of the hypothesisedtransfer period without a falling edge.

FIG. 6 illustrates one embodiment of suitable receiver, i.e. datarecovery, circuitry 600. The input data stream of reversibly encodeddata pulses, Data_down, is input to frequency-locked-loop comprisingfrequency detector 601, adder 602, digital integrator 603, numericallycontrolled oscillator 604 and frequency divider 605. Thefrequency-locked-loop receives the incoming PLM data, Data_down, andgenerates a clock signal, NCLK, at N times the average incoming symbolrate, as detected by say every rising edge of the PLM data. These risingedges will occur at various times within the transfer period, so theapparent input frequency will vary from cycle to cycle, but theintegration in the loop will slow down, and thus attenuate, anyvariation in output frequency, giving an output frequency and hence NCLKclock edge timing which is acceptably constant for recovering data.

The incoming PLM waveform is also received by a sampler 606 which isclocked by the derived clock NCLK and the resulting samples stored inshift register 607. The derived clock signal NCLK is related to themaster clock signal and may be at the same frequency as the master clockMCLK, or may be a multiple thereof, say twice or four times thefrequency to allow oversampling of the input stream, i.e. sampling morethan once every MCLK period to make sure the data is correctly sampled.

The shift register contents, representing the pulse received in onetransfer period, are then sampled in parallel once every transfer periodT_(TCLK), i.e. within every N NCLKs, according to an appropriately timedsymbol sync pulse SSYNC. SSYNC may be generated by a divide-by-N counter608 clocked by NCLK, and reset at the appropriate time by asynchronisation pulse SYNC from synchronisation unit 609.

The sampled pulse data is then fed into a look-up table 610 orequivalent logic processing to derive the payload data. Where, asdescribed previously, the payload data was subject to an XOR combinationwith the control data prior to encoding, the data may re EXOR-ed inlogic units 611 with the recovered Manchester-encoded control data tocompensate. The Manchester-encoded control data CTRL_MCR may beprocessed to recover the control data in a non-Manchester format bybi-phase decoder 612. The data streams may be subject to furtherprocessing, for example to extra left and right 1-bit-PDM channel from acombined data stream or to extract one or more PCM-coded word combinedtogether in a data stream, as previously discussed with relation to anexample of a transmitter.

There are various possible methods to generate an appropriately timedsynchronisation pulse SYNC. For instance the synchronisation process mayrely on receiving one or more symbols transmitted with a uniqueduration. For example with the constraints set out above with an initialperiod of logic 0 in each transfer period and eight time slots there isonly one symbol that can have a duration of 7 time slots, i.e. a datapulse comprising a period of logic 1 for seven time slots. Thus weresuch a symbol to be detected it would be apparent that the rising edgeof the symbol must occur at the end of the first time slot of thetransfer period and the falling edge must occur at the end of thetransfer period. Likewise the set of symbols selected for use may, forinstance include only one symbol with a duration equal to a certainnumber of time slots, say a duration of one time slot—and in which casethe position of the rising and falling edges in the transfer period fora data pulse of one time slot duration will be unique. Synchronisationmay therefore include transmitting a sequence consisting of such uniqueduration symbols to aid synchronisation and initial clock recovery.

Synchronisation may be performed at the start of a data transfer processand then it may be assumed that synchronisation is maintained duringdata transfer unless an illegal symbol is detected. Additionally oralternatively any symbols with a unique duration among the possiblesymbols (or alternatively or additionally a unique rising edge time orfalling edge time) which happen to be transmitted during the datatransfer process may be used to maintain synchronisation. In the eventthat an illegal symbol is detected the receiver may identify thereceived data as containing an error and request re-synchronisation anda resend of the data.

Rather than use a simple divide-by-N counter 608 to generate SSYNC, itmay be preferable to use circuitry with some memory or hysteresis toavoid synchronisation being grossly upset by minor data errors. Forinstance if a SYNC pulse is detected and arrives later or earlier thanexpected, the counter contents may be incremented or decremented by onlyone bit so that the phase of the SSYNC pulse timing is adjusted by onlyone NCLK period at a time, rather than forcing an immediate reset andadjustment by possibly several NCLK periods, due to a possibly mistakenSYNC detection. If the SYNC pulse is detected as arriving only one NCLKperiod early or late, any apparent need for adjustment may be ignored.In some embodiments, for example those relying on transmission ofparticular symbols which may only occur sporadically, or to cope withdata errors causing missing SYNC detection, no adjustment is made if noSYNC pulse occurs for a while, rather than this being regarded as a verylate SYNC pulse occurrence.

FIG. 7 illustrates a possible implementation of a circuit to replace thecounter 608 in FIG. 6. An adder 701 is arranged to add a predeterminednumber P to the output of 1/NCLK time delay 702 every NCLK period, thusincrementing the total by P every NCLK cycle and generating a carrypulse when the adder exceeds full-scale at the adder every Q/P NCLKperiods, where Q is the full-scale of the adder. By appropriate choiceof the value P relative to Q the carry pulse may be used as the SSYNCstrobe pulse.

Adder 703 may add or subtract a further correction amount Corr from thecurrent total, thus advancing or delaying the NCLK cycle at whichfull-scale is reached and thus the timing of the carry pulse generated.This correction amount Corr is generated by comparing the timing of eachdetected SYNC edge with a generated SSYNC pulse using phase detector704. The phase detector output ΔP may be processed by filter block 705before application to adder 703. For example ΔP may be low-pass ormedian filtered to provide a smoother input Corr to adder 703 or may belimited to say +/−1 as discussed above to limit the effect of spuriousor missing SYNC detections.

The embodiment shown in FIG. 6 thus derives, in effect, the master clockand transfer clock from the received data signal. In some applicationshowever the transfer clock may be transmitted on a separate link fromthe transmitter to the receiver. Transmitting the transfer clock doesrequire an additional link, e.g. wire, but this may be acceptable insome applications and it does mean that clock recovery circuitry andsynchronisation circuitry is not required in the receiver, thussimplifying the receiver interface. The receiver can then generate themaster clock signal from the transfer clock.

In general the data receiver circuitry comprises an input for receivinga series of data pulses, and a sampler for sampling each received pulsewithin a transfer period defined by a first clock signal such that thereis a single data pulse with a rising edge and a falling edge in eachtransfer period. The sampler is configured to provide an indication ofwhich of a set of possible data symbols the data pulse correspondsto—based on the timing of the occurrence of both the rising and fallingedges of the data pulse within the transfer period. Decoding circuitrygenerates at least one received digital data stream based on saidindication of which data symbol the received pulse corresponds to. Giventhat multiple data channels are encoded in each symbol for anyindividual channel a plurality of symbols will indicate logic 1 and aplurality of other symbols will indicate logic 0. The decoding circuitryis therefore configured such that a plurality of possible data symbolsmay be decoded as the same value of a data bit of a received digitaldata stream.

Symbol Set Selection

As mentioned above using eight time slots per transfer period may allow28 different symbols to be defined. However, as discussed, not allsymbols may be required in use in some applications, for instance ifencoding 4 bits of data. Therefore some symbols may be used asduplicates for particular data encoding (i.e. a particular data value,say 1101, may be encoded by two or more possible symbols) or somesymbols may not be used at all.

In one embodiment from the set of 28 possible symbols (using eight timeslots—other numbers of symbols would be possible with different numbersof time slots) a minimum set of 16 different symbols may be selected toencode the 4 bits of payload. As mentioned some symbols may be unusedsymbols and removed from the chosen set.

In one embodiment symbols that have both rising and falling edgesfalling in the same half of the transfer period are not used so as toreduce the jitter of the symbol patterns that are input to the clock anddata recovery circuit. In other words the pulse generator may beconfigured such that the rising or falling edge at the start of the datapulse occurs no later than half way through the transfer period and therising or falling edge at the end of the data pulse occurs substantiallyno earlier than half way through the transfer period. The middle columnin FIG. 3 illustrates the remaining symbols with this constraint added.This leaves 19 possible symbols. For 4-bit encoding a further threepossible symbols may be left unused although as mentioned some symbolsmay be used as duplicates.

A total of 17 symbols may be selected as illustrated in the right handcolumn of FIG. 3. In one embodiment symbols 0 and 16 in FIG. 3, i.e. thesymbols illustrated at the top and the bottom of the right hand column,are chosen to be duplicate symbols for the same digital code value. Asmentioned previously such symbols as 0 and 16 may be reserved for timingpurposes and may thus be regarded as ‘special’ Synch symbols—they haveknown fixed position within the symbol timing window (because symbol 0is the only possible symbol with a data pulse duration of 7 time slotsand symbol 16 has been selected to be the only symbol used with aduration of 1 time slot). The use of such unique symbols thereforefacilitates symbol synchronisation. Additionally because they haveminimum and maximum DC values, they may be used to control average DCvalue of the bus, i.e. data link, by control of their selection asrequired.

The DC value of the bus varies with symbols used, but can be controlledby selecting appropriate use of the equivalent value 0 and 16 symbolsusing a simple DC level monitoring circuit.

Other symbols sets may be chosen however which optimise various aspectsof the data transfer performance such as minimising any DC imbalance onthe PLM data line. As mentioned the symbol set used may be selected witha view to reducing jitter. FIG. 8 illustrates an alternative symbol setusing just 16 symbols.

Point-to-Point/Multi Point Implementation

Embodiments of the present invention can be used for one-waypoint-to-point data transfer from a transmitter device to a receivingdevice as shown in FIG. 2. Two-way, i.e. bi-directional, data transfercould also be implemented with separate transmitters and receivers. Inthis implementation each may use its own master clock.

Such a bi-directional link could be used to transfer multiple datastreams, e.g. left and right audio data, over a single wire to aperipheral device such as a headset or the like where the data streamscan be decoded locally and provided to the appropriate components i.e.transducers.

In some embodiments the PLM signal may be split into two or more signalsbefore decoding such as illustrated in FIG. 2b . FIG. 2b shows atransmitter 201 having a modulator 204 such as described abovetransmitting a PLM signal to two receivers 202. Most of the PLM signalis transmitted over a single link 203 but at some point 205 the PLMsignal is tapped 206 so that both receivers 202 are provided with aversion of the PLM signal. FIG. 2b thus shows separate receivers 202 fortwo audio components, each associated with a DAC-amplifier 206 andspeaker 207. In a device having internal stereo speakers the speakersmay typically be separated and arranged to receive signals from an audiosource such as codec and thus the arrangement shown in FIG. 2b may bepreferred. In this case the receivers may be integrated with therespective DAC-amplifier.

The extract circuitry of each separate receiver may receive a logicsignal to denote whether to use left or right channel data. In FIG. 2bthis is shown as a pin which is pulled high (V) or low (GND). In otherembodiments this logic signal may be derived from some non-volatilememory or circuit link that is programmed during chip manufacture andtest, or by designing different left- and right-channel variants of thecircuitry, for instance the left-channel may be co-integrated with othercircuitry, whereas the right-channel may be some separate simplerdevice, expected to be located at some distance.

Bus Implementation

In one embodiment however devices may be arranged in chain configurationas illustrated in FIG. 9. As shown in FIG. 9 there may a master device901 which acts as a bus master.

In one embodiment, such as shown in FIG. 9a each device has two pins—PLMreceive (alternatively termed PLM-down or just Data-in) and PLM transmit(alternatively termed PLM-up or just Data-out). The bus master transmitsdata for the devices on the chain to the first slave 902 in the chain bytransmitting appropriately encoded symbols using the protocol describedabove. This first slave will receive the PLM data stream, and recoverthe master clock signal from the data. The first slave will alsodetermine the appropriate symbol and decode the data. The first slavewill then re-encode the data (with any modifications as necessary—forinstance updating an address nibble field) using the recovered masterclock signal and transmit an appropriate symbol to the next device inthe chain. Each slave will recover the master clock signal, decode thedata, modify as appropriate and re-encode the data until the data isreturned to the bus master.

Typically if a device in the chain is disabled it must still decode andre-transmit the incoming stream, updating the address fieldappropriately. Thus each device in the chain will add at least onetransfer period in latency to the chain. In other words the slave mustreceive the symbol and decode the data and re-encode the data asubsequent transfer period.

However in some embodiments if a particular slave is not be used at alla bypass may be enabled where the data pulses are simply retransmittedbefore being decoded. This avoids introducing any delay other than aninevitable small propagation delay. Activation and deactivation ofbypass mode may be implemented via some other control link or via thestandard control frame. If each receiver has a fixed address then thecontrol frame may be used to instruct a receiver to enter bypass modewithout any impact on the addressing of subsequent receivers. In whichcase the bypassed receiver may forward the symbol without anymodification but subsequently decode it to read the control frameinformation (the clock recovery circuits will also thus continue tooperate to maintain frequency lock and synchronisation). Alternativelybypass may be enabled at bus initialisation stage and only rescinded byre-initialising the bus, in which case a bypassed device may effectivelybe disabled until the bus is re-initialised.

In other embodiments however the symbol clock, i.e. the transfer clocksignal, may be provided to at least some of the slave devices directlyas indicated in FIG. 9b . In this embodiment the devices may have threepins: Data-in, data-out and clock (which is clock-out for the master andclock-in for the slaves). The slaves can generate the master clocksignal from the transfer clock signal as described above.

In this embodiment there is a single bus, i.e. link, for data transferin one direction which returns to the bus master. In other embodimentshowever there could be two buses for passing data in opposite directionswith an appropriate bus master at the relevant end of each bus.

The chain topology facilitates ‘consume and replace’ functionalityChannels can be reused if a consuming device (e.g. a speaker driverwhich pulls data off the interface) is placed before a generating device(e.g. a microphone which pushes data onto the interface) in the chain.

FIG. 10a illustrates an example of a configuration of a slave device1000 that is both a consumer or sink of audio data, i.e. has a speaker1001, and acts as a generator or source of audio data from a microphone1002. The PLM data down signal is received from the previous device inthe chain which may the bus master or a preceding slave device. The PLMdata signal is received by clock recovery circuitry 1003 to generate amaster clock signal as described above. A PLM decoder 1004 then decodesthe data in each of the data channels and the control channel (ifpresent). If one of the data channels, say data channel 2, containsaudio data for driving the speaker 1001 of that slave device therelevant audio data may be extracted and used to drive the speaker. Theaudio data on the other channels, e.g. data channels 1 and 3 is passedto the PLM transmitter 1005 for re-encoding. The relevant data channelused for that slave device may have been communicated previously viasuitable control data or it may be predefined.

Data from the microphone 1002 may also be used to encode audio data foronward transmission to the next device in the chain. The audio data maybe encoded on any space on any of the data channels. In this exampledata channel 2 is used to transmit data to the speaker of the slavedevice and thus may be free for use to encode the microphone data. ThePLM transmitter thus encodes the data from data channels 1 and 3together with the data on channel 2 from the microphone and any controldata and generates and appropriate symbol for onward transmission. FIG.10b illustrates another embodiment of a slave device 1000 b thatillustrates the control data processing and thus includes control datacircuitry 1006 including a control slot decoder 1007 and circuitry 1008for controlling operation of the audio component based on the controldata.

In order for the bus master to communicate with the various slavedevices the slave devices are given addresses. In one embodiment themaster may be arranged to be address 0. Slave device addresses may beincremented by one for each subsequent device on the bus. The addresscan be carried in a nibble (4 bits) in a byte (8 bits) of the 32 bitcontrol word described above. Up to 15 devices can be addressed in thisway using the embodiments described above.

The control frame may include a BUSY bit field which the master can useto indicate that the frame contains valid data. If the BUSY bit isasserted then the control data in the frame is not altered (apart frompossibly an address nibble field or the attention bit ATTN discussedbelow) If a slave recognises that the BUSY bit is de-asserted it is freeto provide data in that frame. If a slave does use a frame and adds datait can assert the BUSY bit so that the data is not overwritten by asubsequent slave in the chain.

The control frame may also comprise an attention bit field ATTN which aslave can use to get the attention of the master. In one embodiment onlyone slave at a time can assert the attention bit, in which can it mayalso add some data such as its address and maybe an error code (in whichcase it also asserts the BUSY bit to ensure such data is delivered tothe master). Thus a slave must wait for a frame where the attention bithas not been asserted by a previous slave (and, in order to transmitdata, such a frame where the BUSY bit is not asserted). This means thatearlier slave devices in the chain effectively enjoy higher priority andthe master should preferably include regular frames without the BUSY bitasserted. In another embodiment any slave can assert the attention bitat any time. The master, on detecting the attention bit has beenasserted will poll the slaves to find out which slaves requireattention.

The control frame may also comprise a command bit field CMD. Its purposeis to allow the Master to send a “command” to a Slave, eitherindividually or in a broadcast fashion. The “command” may beencapsulated in those control bits that follow the Slave Address bits inthe frame, e.g. D[15:0]. When the CMD bit is asserted, the “command”must be treated as an instruction by each slave. Bus critical “commands”like “bus reset” benefit from the added protection of a wide instructiondecode, prior to execution.

The master typically is the bus clock master and devices may recover aclock from the bus as discussed previously. A bus master has the MCLKavailable so no clock recovery is required. However data recovery mayrequire allowance for uncertainty in the total propagation delay alongthe chain.

As previously discussed there may be three levels of synchronisation inthe system: MCLK frequency and phase lock; symbol (phase); and frame.

Bus Initialisation

Bus initialisation typically involves a series of synchronisation steps.Initially each source of audio data on the bus will be deactivated fromtransmitting—i.e. in the off state each device transmits zero.

To start the synchronisation process the master may synchronise with thefirst slave device on the bus using the synchronisation processdescribed above. Thus the master may send a sequence of synchronisationsymbols onto the bus to initialise clock recovery circuitry.

Once the first device in the chain has locked onto the incoming datastream and attained bit and symbol synchronisation it will starttransmitting the same sequence to the next device in the chain.

When the master receives this sequence from the last device in the chainit is then aware that the chain has successfully synchronised. At thispoint data transmission can commence.

A timeout may be used to prevent lock-up should there be asynchronisation error.

During initialisation a device must receive a number of alternatingsynchronisation symbols before it starts transmitting.

Initialisation time from a cold start is likely to be dominated by thetime to frequency lock of the clock multiplier used to recover MCLK indevices.

Slave Device Address Enumeration and Addressing

In some embodiments each slave device is given an address based on itsposition on the bus. Thus initially during reset the slave devices mayset their slave addresses to a known value, such as zero, which is not avalid slave address, e.g. it may be reserved for the bus master. Afterthe Master has received confirmation of ‘Bus Initialisation’, it maythen assign an address to each Slave via the control channel.

In one embodiment the bus master may transmit a frame with a BUSY bitset zero and a slave address nibble field set for the target slave. AnATTN bit, which is used by a slave device to flag a status to the masteris also set to zero. The master then waits for this frame to bereturned, with the BUSY and ATTN bits set to one before transmittinganother such frame.

Each slave address, on start up or reset will have an initial addressset to a forbidden value such as zero. In such a state it may be ableonly to propagate data that it receives and can do nothing else untilgiven an address by the master.

The slave device thus waits until it receives a frame with the BUSY bitset to zero and the ATTN bit also set to zero (indicating no previousslave has made use of the address data. When such a frame is receivedthe relevant slave consumes the address field data as its own addressand then retransmits the frame with the BUSY and ATTN bits set. Allsubsequent slave devices will simply retransmit this frame which willthus be passed back to the master.

In some embodiments, the Slave may insert information about itscapabilities into the frame it transmits. This capability “signature”may comprise a manufacturer or device type identifier and indicators asto whether the Slave can for example recognise PCM or multiplexed stereodata, whether it has a sleep mode, how many channels of input or outputdata it can accept or deliver etc.

In this way all slave devices on the bus can be set up with a uniqueaddress.

Once the bus has been enumerated and each Slave knows its address, theaddress space may be used to identify payload destination. Ifimplemented as an addressable register, a Slave can be re-enumerated bythe Master at any point by writing zeroes to the Slave Address Registerand then assigning the address. An obvious restriction is that alladdresses assigned by the Master must be unique for each Slave, andmust, in such an embodiment, have a non-zero value.

Crash Recovery

If any of the devices in the chain detect an invalid signal bit orsymbol it may be assumed that synchronisation has been lost. Therelevant slave device needs to notify the bus master thansynchronisation has been lost. In one embodiment each slave devicereceiving an invalid symbol may simply propagate that symbol to the nextdevice so it is received back at the bus master. The relevant slave mayalso assert an ATTN attention bit in the control frame as discussedabove.

When the master detects the invalid symbol or an incorrect slave addressor the like it may initiate a crash recovery procedure. This may involvethe master attempting to read status information from each of the slavedevices. The master may re-initialise the bus using the samesynchronisation steps as described previously. However the master mayonly re-initialise the bus after a certain number of errors are detectedand if the error is in an audio data stream it may simply be ignored (asthe corrupted audio data may not be readily replaced).

It should be noted that frequency lock is maintained during a crashcondition therefore recovery is rapid. (PLLs should be designed to‘hold’ frequency during crash)

Error Detection

Worst case link failure is anticipated to cause loss of one complete 64symbol frame. This is the same magnitude of failure as a single failureon a bus such as I2S or HDA link.

Embodiments of the present invention may thus use a pulse lengthmodulation (PLM) protocol to combine multiple data streams, e.g.multiple audio streams and/or control data into a single PLM stream,i.e. the data pulses recoverably encoding the data, at a PLM datafrequency (i.e. transfer period clock rate), which may be a multiple ofthe underlying audio data frequency, for instance 64, 128 or 256 timesthe audio data frequency. Both clock and data are embedded into thesymbol stream and may be transmitted on a single link, e.g. a singlewire or single bus. The physical interface is thus simpler than manyknown transfer protocols and the protocol allows multiple data streamsto be transmitted at a high data rate. As the data can be transmittedwith an underlying PDM encoding the latency in data transfer is very lowwhich may be particularly useful for applications such as ambient noisecancellation when the time allowed to received audio data, applyappropriate processing, and transmit appropriate cancellation data isvery short. The data streams transmitted may be oversampled datastreams, i.e. data streams at sample rates which are significantlyhigher that the underlying audio sample rate. For instance with 48 kHzunderlying audio sample rate the data streams may correspond to anoversampled data rate of at least 700 ks/s or at least 3 Ms/s.

The clock is effectively sent as the symbol rate. As mentioned this maybe a multiple of the input sample rate, e.g. 64 time the audio samplerate (fs) for a 48 kHz audio sample rate.

The data is sent encoded by the pulse length and position in thetransfer period, i.e. by the selection of the symbol. 4 bits may be sentper symbol, e.g. 3 data bits and 1 control bit (or in some embodimentsthe control data may be bi-phase encoded as will be described so thereis only a half bit for the control channel per symbol), using 8 possibleclock edges. For a symbol frequency of 3.072 MHz this requires a clockfor generation of the symbols of around 24.576 MHz.

A symbol has one rising and one falling edge. Power consumption and EMIcharacteristics of the bus are therefore dominated by a fixed edge(symbol) rate (typically therefore 3.072 MHz).

The protocol may be implemented on a single wire bus. In one embodimentthere are 3 data channels per bus supporting up to 3×64 fs PDM channelsper bus (more PDM channels at reduced rate can be supported byinterleaving across alternate symbols if required).

There may additionally be one control channel per bus.

In general then embodiments of the invention enable transfer of multipledata streams of encoded data over a single wire link which allows clockrecovery by receiving devices. Embodiments may also provide both encodedclock and data on a single wire. Embodiments support PDM and PCM datastreams, such as audio data streams, and/or control data. In a busimplementation there may be simple address enumeration due to a chaintopology and a simple physical interface—one digital output and onedigital input per master or device so 2 pins. There may be, in someimplementations up to one master and 15 devices on a bus.

Fixed Rising/Falling Edge

The embodiments described above modulate the time of the rising edge andthe falling edge of the data pulse in the data transfer period toprovide the set of possible symbols to encode the data. In analternative aspect however the symbol set may be constrained so that oneof the rising edge or falling edge always occurs at the same time in thetransfer period. This additional constraint reduces the number ofpossible symbols (for a given number of time slots) but makes clockrecovery more straightforward. For example consider the use of eighttime slots in a transfer period, if the set of possible symbols isconstrained to include only those symbols where the rising edge is atthe end of the first time slot (to ensure a gap between data pulses)there are only seven possible symbol lengths available. In thisembodiment as the position of the rising edge in the transfer period isfixed the data is encoded through the duration or length of the datapulse. This means that the time between the rising edges (or fallingedges) of successive data pulses are always separated by a period equalto the transfer period, i.e. the rising (or falling) edges can be usedmore directly as a clock signal a frequency equal to the transfer clockfrequency. This can simplify clock recovery and there is no need tosynchronise to a transfer period (in effect the fixed rising or fallingedge of a data pulse defines the start of a transfer period).

In this embodiment, referring back to FIGS. 2 and 2 b, audio interfacecircuitry of the first component 201 has a pulse-length-modulation (PLM)modulator 204 which receives audio data input data, PDM-R and PDM-L, forthe left and right audio channels. In this example the L/R input dataare separate 1-bit (i.e. PDM) digital input data streams for eachchannel, i.e. left channel and right channel. At regular intervals, forinstance in each period of a first clock signal or transfer clock signalTCLK (which preferably matches the sample rate of the L/R 1-bit audiodata input) the PLM modulator generates a data pulse with a length thatdepends on the logical combination of L/R input audio data. For left andright 1-bit audio data inputs there are four possible combinations ofinput data and the PLM modulator 204 generates a data pulse having alength (i.e. duration) which varies according to the output of theparticular L/R input data combination, as illustrated in table 2 below.

TABLE 2 PDM Combined Pulse length PDM-L Value PDR-R Value value(T_(MCLK)) 0 0 00 1 0 1 01 2 1 0 10 3 1 1 11 4

As described above the pulse length is preferably determined as amultiple of a clock period of a second clock signal, or master clocksignal, MCLK, which has a frequency which is a multiple of the transferclock signal TCLK. It will be clear that the frequency of the masterclock signal MCLK should be at least four times the frequency of thetransfer clock signal TCLK and is preferably at least five times thefrequency of the transfer clock signal TCLK to allow for a pulse lengthof 4T_(MCLK) within each period of the transfer clock signal TCLK andalso allow a gap between pulses to ease downstream clock recovery. Inother words with five time slots the maximum pulse duration is at leastone time slot less than the transfer clock period to allow at least onetime slot gap between successive pulses.

FIG. 11 illustrates the transfer clock signal TCLK and master clocksignal MCLK, which in this example has a frequency five times that ofthe transfer clock signal TCLK. FIG. 3 also illustrates that in eachperiod of the transfer clock signal TCLK the right and left input datastreams PDM-L and PDM-R will each have a data value of 1 or 0 dependingon their respective current data i.e. logic levels. The resulting PLMdata signal (PLM), generated to represent a logical combination of L/Rinput audio data as per table 1, is also illustrated. This comprises aseries of data pulses, at the frequency of the transfer clock signalTCLK, where the length of each individual data pulse encodes a datavalue for the left audio channel and also a data value for the rightaudio channel.

It will be noted that there is always one data pulse per period of thetransfer clock signal TCLK and that the data pulses are arranged so thatthere is always a gap between data pulses. This means that the PLM datasignal can itself be used to recover the transfer clock signal TCLK bythe receiving interface circuitry. In the example shown, the risingedges of the clock signal are separated by the clock period but theskilled person will appreciate that the falling edges could instead besynchronised to the transfer clock signal TCLK.

The receiver can extract the transfer clock signal TCLK from the PLMdata signal. As mentioned above the period between the rising (oralternatively the falling) edge of each data pulse indicates the firstclock period. Each receiver then determines the length of each datapulse and uses this to determine the relevant audio data for theappropriate device. From table 2 above it can be seen that pulse lengthsof 1 or 2 times the master clock period indicate a data value of 0 forthe left channel and pulse lengths of 3 or 4 times the master clockperiod indicate data 1 for the right audio channel. Given that themaster clock signal frequency f_(MCLK) is a known multiple of thetransfer clock signal TCLK frequency f_(TCLK) the relevant length of thedata pulse can therefore be determined.

Each receiver therefore derives a version of the transfer clock signalTCLK and a digital data signal for the relevant audio channel. Thedigital data signals may, for instance by a 1-bit PDM signal but it willbe appreciated that other formats are possible.

The transmitter 201 is therefore a PLM encoder for receiving data andproducing a PLM data signal encoding such data. The receiver 202 is aPLM decoder which receives the PLM data signal and decodes the signal.

FIG. 11 illustrates that the data signals PDM-RX and PDM-LX,respectively extracted by the receivers) 202 are the same as therespective input data PDM-R and PDM-L.

This embodiment of the present invention therefore supplies simultaneousaudio data for multiple audio channels using only a single communicationlink, i.e. a single data wire, without requiring any clock signals to besent.

FIG. 12 illustrates one example of a suitable PLM modulator 204, i.e. aPLM encoder, that may be used as part of an audio interface transmittingdata pulses with fixed rising edges. As illustrated the PLM modulatorhas a combiner 1201 that receives the two input 1-bit data streams,PDM-L and PDM-R. The combiner 1201 converts the input data to a 2-bitcombined data stream PDM-C (as described above in relation to table 2).A counter 1202 is also arranged to receive the transfer and master clocksignals TCLK and MCLK respectively. The master clock signal MCLK isarranged to clock the counter to increment and the transfer clock signalTCLK is provided at the reset input. Thus each period of the transferclock signal TCLK the counter will increment at the rate of the masterclock signal MCLK. The output is thus is a sawtooth ramp waveform thatis reset to zero at the start of each period of the transfer clocksignal TCLK (or alternatively the counter could be arranged to countdown from a certain starting level each period).

The output of the combiner 1201 and counter 1202 is compared bycomparator 1203 which is clocked at the second clock frequency MCLK, andis configured to give a high output if RAMP is greater than the combinedsignal PDM-C. At the first relevant (e.g. rising) clock edge of MCLKafter the counter has reset, the RAMP signal will be zero, and thecombined signal will be zero or greater, so the output of the comparatorgoes high at the beginning of the first relevant edge of MCLK in eachperiod of the transfer clock signal TCLK. At successive MCLK edges, theRAMP signal will have increased: the comparator output goes low when theramp signal has exceeded the combined data value. This results in a datapulse where the rising edge is synchronised to the transfer clock signalTCLK (i.e. the rising edge of each data pulse occurs at the firstrelevant MCLK edge after the rising edge of the TCLK clock signal) andthe pulse length depends on the combined data value. It will of coursebe appreciated that the comparator could be arranged so that the outputgoes low at the start of the clock period and goes high only as a resultof the comparison to synchronise the falling edges of the data pulses.

FIG. 13 illustrates the various data waveforms and shows how thecomparison of the combined data value PDM-C with the ramp signal cangenerate the PLM data signal.

It should be noted that the PLM modulator described with reference toFIG. 12 is described as acting on 1-bit PDM input data streams. In someembodiments a received audio data stream for an audio channel may beconverted into a 1-bit PDM data stream before it is input to the PLM.For example if input analogue audio data from a microphone or otheranalogue source is received there may be a need to convert it to a 1-bitPDM digital data stream. Alternatively if audio data is received orstored in multi-bit PCM format the audio data words may be convertedinto a 1-bit PDM stream using known conversion techniques. Thusreferring to FIG. 2 first component 201 may comprise conversioncircuitry (not shown) for converting an input audio stream into a 1-bitPDM stream which is then provided to the PLM modulator.

However in some embodiments the PLM may be arranged to receive multi-bitinput data for one or more audio data channels. For example considerleft and right stereo data where each channel comprises two-bit digitalaudio data. This means that each clock period there may sixteen possiblecombinations of audio data. If the master clock signal MCLK is arrangedto have a frequency which is at least seventeen times that of thetransfer clock signal TCLK the sixteen possible combinations maytherefore be encoded by sixteen different pulse lengths. The principleof operation is the same, for example a four-bit combined data signalrepresenting the state of both input channels may be produced andcompared to a counter ramp that increments at the rate of the masterclock signal MCLK. Clearly this could be extended to digital signals ofhigher numbers of bits but with a consequent increase in the number ofdifferent pulse lengths required and a corresponding increase in thespeed of the master clock signal MCLK compared to the transfer clocksignal TCLK, which may require very fast components and accurateresolution between relatively small differences in pulse length. Anotherexample is 3-level data as used for control of Class-D H-bridge outputstages, requiring nine possible pulse lengths for a stereo pair.

It will of course be appreciated that data transmitted by the PLM datasignal could be PCM encoded or use some other form of data words in asimilar fashion to that described previously if a frame or data wordsynchronisation could be sent, for instance by increasing the number oftime slots to send a control data channel in addition to the audio datachannels or using a pulse of a given length which does not encode databut does provide synchronisation.

It will also be noted that other techniques for pulse length modulationin general, for example using delay-locked loops (DLL) are known andcould be adapted for use in embodiments. Also it should also be notedthat the transfer and master clock signals could be received by the PLMmodulator or one or both of these signals may be generated by the PLMmodulator. For instance the PLM modulator may receive the transfer clocksignal TCLK, for example as a clock accompanying and synchronous to thePDM data and generate the master clock signal MCLK or the PLM modulatormay receive no clock signals and may recover the transfer clock signalTCLK from the input data using known clock recovery techniques.

FIG. 14 illustrates one embodiment of suitable receiver, i.e. a PLMdecoder for use in the embodiment using a fixed rising edge. In thisembodiment, the clock signal is recovered using a Delay-Locked-Loop(DLL) comprising a voltage-controlled delay line 1401, itself comprisinga chain of voltage-controlled delay elements 1402, together with aphase-frequency detector (PFD) 1403, charge pump 1404 and loop filter1405 as known. The DLL receives the PLM data signal and the PFD 1403 isdesigned suitably so that each rising edge of delay line output SYNC islocked to a rising edge of the incoming PLM data stream. Thus the delayalong the delay line becomes equal to a period of the original TCLKclock.

The duty cycle of SYNC will vary with the data carried on PLM data. Toestablish a clock with 50% duty cycle, SYNC is applied to the set inputS of an edge-triggered RS flip-flop 1406, while a signal y5 from halfway along the delay line is applied to the reset input.

To extract a measure, PL, of the length of each transmitted PLM datapulse (i.e. each pulse of data signal PLM Data), the data at a number ofequally spaced taps of the delay line 1401 is summed in a summer 1407.The delay taps are arranged to sample the signal at intervals equal tothe second clock frequency MCLK and in sufficient number to be able todiscriminate between the number of different possible pulse lengths. Inthis embodiment, where the second clock frequency is known to be fivetimes the frequency of the transfer clock signal TCLK and there are fourpossible different pulse lengths, there are five delay taps at equalspacings along the delay line.

The determined data pulse length value PL is then applied to a look-uptable 608 or equivalent logic to decode the PLM Data signal in therelevant PDM audio data channel or channels. In the embodimentillustrated in FIG. 2b , where there is a different data extractionmodule for each data channel, respective logic inputs LR are tied toground or VDD to signify left or right channel. This signal LRrepresenting whether PDM-L or PDM-R is to be extracted may also beapplied to the look-up table to determine the appropriate data value.Alternatively a multiplexer between the MSB and LSB of the two bit PLdata value, driven by LR, may be used to select the appropriate dataoutput. In other embodiments, especially with more than two receivingchannels, alternative methods of identifying each receiver may beimplemented and used to extract appropriate data using suitably adaptedtruth tables.

The look-up table contains a truth table such as shown in table 3 below.

TABLE 3 PL LR PDM-L PDM-R 00 0 — 0 01 0 — 1 10 0 — 0 11 0 — 1 00 1 0 —01 1 0 — 10 1 1 — 11 1 1 —

It will be appreciated however that in other embodiments a single dataextraction module may be used to extract both the PDM-R and PDM-Ltogether.

It will be noted that this detection method does not require explicitrecovery of the master clock signal MCLK (or any clock signal fasterthan the first clock signal). However if this is required, for instancefor other circuitry, such a clock may be generated by logicalcombination of the tapped outputs y1 to y9 as known. The recovered firstor second clocks may be used for other functions in the device. Indeedthe rising (or even both) edges of the unprocessed MPDM signal may beused as a clock edge.

FIG. 15 illustrates, for an embodiment with a fixed rising edge, the PLMData input signal that would be received for the various possible datacombinations, how the transfer clock signal TCLKX can be recovered andhow the output of the delay tap y5 can set a 50:50 duty cycle for theclock signal. These waveforms also shows how the outputs of the delaytaps y1 to y9, when summed, provide an indication of the length of thedata pulse which can then be used to extract the PDM-R and PDM-L audiodata streams.

In correct operation, the sampled output of the first delay tap y1 isalways 0 and the sampled output of the last delay tap y9 is always 1.This can be used to generate a flag indicating correct operation i.e.lock of the DLL loop and correctly formatted input data.

In this case it can then be noted that the value of PL only varies dueto the contribution of delay taps {y3, y5 and y7}. If the value PL wereproduced by summing just the output of these three delay taps then thevalues of PDM-L and/or PDM-R can then be selected merely as the MSB andLSB bits of PL, as illustrated in Table 3.

The embodiments described above with reference to FIG. 12 relate to twochannels of audio data, e.g. left and right stereo data. However it willbe appreciated that the principle of PLM encoding with a fixed rising orfalling edge position in the transfer period can be extended to morechannels of audio data, for example for surround sound. For example ifthere were four channels of audio data, left-front, left-rear,right-front, right-rear and each data channel was a 1-bit PDM datastream then the combined data would comprise a 4-bit data signal whichcould be encoded by 16 different pulse lengths.

In the embodiments described above the number of possible pulse lengthsthat can be produced by the PLM modulator has been set to simply provideunique recoverable encoding for each possible input data combination,i.e. the period of the data transfer clock signal TCLK has been dividedinto an equal number of time slots sufficient to allow unique encoding.However in some embodiments of the invention the number of possible timeslots may be greater than is required just for unique coding of theaudio data channels. These extra slots can be used to provide a numberof additional features and advantages.

In one embodiment the PLM modulator may be arranged so that at least onecombination of input audio data for the left and right audio channelscan be encoded as at least two different alternative data pulse lengths.In other words a particular combined data value, say 10, may be encodedas a first pulse length or alternatively as a second, different, pulselength.

Varying the pulse lengths can help reduce problems with EMI byeffectively whitening the data signal spectrum. Thus the PLM modulatormay be arranged to use the first pulse length to encode one instance ofthe combined data value and the second pulse length to encode a laterinstance of the same combined data value. In other words the PLMmodulator may vary between the two different alternative data pulselengths when encoding a given combination of input audio data. In oneembodiment the PLM modulator may simply alternate between the possibledifferent encodings but in other embodiment the PLM modulator may bearranged to select between the alternative encodings at random.

In one embodiment the PLM modulator is arranged to vary between thepossible encoding pulse lengths so as to control the average pulselength of all instances of a particular combination of input data.

Advantageously for any possible combined data values that havealternative possible encodings the possible encodings may be symmetricabout a given pulse length. FIG. 16 illustrates a period of the transferclock signal TCLK which is divided into eight separate time slots, i.e.the frequency of the master clock signal MCLK is set to be eight timesthat of the transfer clock signal TCLK so that there are seven possiblepulse lengths (it will be recalled that a pulse length using all eighttime slots is not used to ensure a gap between data pulses, to allowsimple clock recovery using the rising edge thus available at each firstclock period). The input data combination 11 (i.e. data 1 on the leftchannel and data 1 on the right channel) may be encoded as a pulselength of four time slots (illustrated as position 3). This encoding maybe the only option for an input data combination 11. All the otherpossible input data combination values, 10, 01 and 00 however may eachbe encoded as two possible pulse lengths. In each instance, asillustrated, the pulse lengths may be symmetric about a given pulselength. Thus input combination 00 may be encoded as a pulse of one timeslot in length (position 0) or 7 times slots in length (position 6).

In this example the various alternative pulse lengths are symmetricabout a pulse length that is half the interval between pulses (which maygive advantages for DC balance in some implementations). However thisdoes not need to be the case and where there more time slots availableany desired pulse length may be chosen that has sufficient time slots oneither side.

The truth table that the PLM modulator uses to encode the input data topulse length may therefore be based on that set out in table 4 below.

TABLE 4 Left data Right Data R value position 0 0 0 0 0 1 0 1 1 0 0 2 11 0 3 0 0 1 6 0 1 1 5 1 0 1 4 1 1 1 3

The R value is used to control which alternative encoding is used forthose input data combinations which have alternative encodings. If arespective R value is simply alternated between successive occurrencesof each input code pair, the average pulse length will be constant,irrespective of the input data.

It may be good enough merely to alternate a common R in alternate clockcycles, assuming random input codes. However in some embodiments the Rvalue for each input code pair or a common R value is generated as arandom (or pseudo random) sequence. This ensures that the output signalspectrum is whitened.

The receiver would need to be able to distinguish between sevendifferent pulse lengths and thus could be implemented with a delay linehaving 16 elements and 8 taps. A different look-up table would alsoclearly be required.

In another embodiment redundant time slots may be used to encode atleast one additional data channel or side data channel. In other wordsthe PLM modulator may be arranged so that at least some data pulsesencode not only multiple audio channels but also at least one additionaldata channel. The additional data channel may, for instance comprisecontrol data for controlling aspects of the audio apparatus. Forinstance audio control data for enabling or disabling mute or one ofseveral standby or low-EMI modes, or controlling volume may betransmitted.

The additional data may be encoded as a separate data channel asdescribed above. Thus with two 1-bit PDM audio data channels, e.g. leftand right audio data, and a 1-bit control data channel, the eightpossible unique data combinations of audio and control data could beencoded as eight different pulse widths, requiring the clock period tobe divided into at least nine separate time slots. Thus the frequency ofthe master clock signal MCLK could be set to be at least nine times thatof the transfer clock signal TCLK.

With a control channel all the features described above with regard tothe use of frames and control of multiple devices on a bus can beimplemented using this fixed rising or falling edge embodiment. Controldata could provide frame synchronisation allowing data to be encoded asdata words.

However whilst it is certainly possible to derive a master clock signalMCLK to provide any desired number of time slots in the period of thefirst time clock signal it is noted that it most convenient to generate2^(N) time slots, i.e. to derive a second clock frequency which is 2^(N)times the first clock frequency. This would provide 2^(N)−1 possiblepulse lengths for an embodiment with a fixed rising or fixed fallingedge position of the data pulse. In other words constraining the risingedge (or alternatively the falling edge) of the data pulse to a set timein the transfer period limits the number of possible symbols. Thusproviding sufficient time slots to encode both the two audio datachannels and the control data channel in each data pulse, withconvenient generation of the master clock signal MCLK and a fixed risingor falling edge time, would involve generating 16 (i.e. 2⁴) timeslots—with the associated requirements for fast circuitry.

It has been realised however that in some embodiments the data raterequired for additional data such as control data may be lower than thatrequired for the audio data channels and it may not be necessary to haveframes defined by the control data. Thus in one embodiment some, but notall, data pulses may also be encoded with additional data. Provided thatthe number of possible pulse lengths is at least one greater than theminimum needed to uniquely encode the audio data channels then at leastone particular input combination of audio data can be encoded in twoalternative ways and any instances of such audio data combination canthus be used to transfer the additional data.

Thus consider two 1-bit PDM audio data channels, for instance left andright stereo audio data. As described previously a combined signalrepresenting both audio channels will thus be a two bit signal and willrequire four possible pulse lengths for unique encoding of the audiodata. This would require at least five time slots in each period of thetransfer clock signal TCLK. As also mentioned it may be more convenientto generate eight time slots (i.e. 2³ time slots) rather than five, thusproviding seven possible pulse lengths. As only four pulse lengths areneeded to encode the audio data this leave three possible pulse lengthsthat can be used as alternatives encodings for the audio data. Thismeans that three out of the four possible audio data combinations can beused to transmit additional data.

For example encoding could be performed according to the followingtable:

TABLE 5 Control/side Pulse length Left audio data Right audio Datachannel data (T_(MCLK)) 0 0 0 1 0 1 0 2 1 0 0 3 1 1 N/A 4 0 0 1 7 0 1 16 1 0 1 5 1 1 N/A 4

Thus a pulse length of 1T_(MCLK) indicates an input audio datacombination of 00 and a control data value of 0 whereas a pulse lengthof 7T_(MCLK) indicates an input audio data combination of 00 but acontrol data value of 1. However the audio input data combination 11 isonly be encoded as one possible pulse length, equal to 4T_(MCLK).Therefore any instances of the input audio data combination 11 meansthat the particular data pulse can not also be used to encode thecontrol data.

Assuming the input audio data combinations provide effectively randomsequences this still allows an average of 0.75 control bits per symbolof the MPDM data which is sufficiently high for most control data,although clearly there is a code dependent latency. This code dependencemay be mitigated by re-coding the audio input data stream to reduce thedensity of 11 states by known methods, for example bit flippingtechniques.

FIG. 17 illustrates an embodiment including control data transfer. Inthis embodiment PLM modulator 1701 receives left and right audio data as1-bit PDM signals as described previously but also receives control dataCDATA. The PLM modulator 1701 is responsive to transfer clock signalTCLK and master clock signal MCLK which is eight times the frequency ofthe transfer clock signal TCLK. The PLM modulator may implement theencoding shown in table 5 above.

In one embodiment the audio data may be combined together with thecontrol data value to provide a pulse length data value in accordancewith the table 5 above. The pulse length data value may be compared witha sawtooth ramp signal that increase each time slot as describedpreviously.

The data extraction modules 1702 and 1703 operate as describedpreviously to extract the clock and relevant audio data signals exceptthat the truth tables include indications of the value of control datafor all pulse lengths other than 4T_(MCLK). The extracted control datacan then be used to control DAC-amplifiers 1704, 1705 respectively.

In some embodiments control data may only need to be transmittedinfrequently. When no control data needs to be transmitted the controldata signal could be effectively be set to be a constant string of zeros(or ones). However in one embodiment when there is no control data to betransferred the PLM modulator 1701 may be arranged to vary between thealternative encodings for the audio data input combinations as describedpreviously with respect to FIG. 16 in order to whiten the signal dataspectrum and provide the EMI benefits mentioned above.

The PLM modulator may therefore be arranged such that, when no controldata is present, those audio data combinations that can be encoded byalternative pulse lengths are varied between the alternativeencodings—either alternated or selected randomly as noted previously.Advantageously, as previously noted the two possible encodings for agiven audio combination have pulse lengths which are symmetrical about agiven pulse length. When subsequently control data is available the PLMmodulator may then encode the control data as described.

In this arrangement the receiver will need to be aware whether thevariation between two alternative possible pulse lengths for a givenaudio combination is simply random variation for the purposes ofreducing EMI issues or whether the encoding represents control datawhich has been encoded into the data pulse.

Therefore in this embodiment there is at least a first reserved sequenceof control data modulation which the PLM modulator is arranged not touse when not transmitting control data. The receiver may decode eachdata pulse as if it were encoding control data and look to see whetherthe reserved control data pattern occurs. If not the receiver willignore the control data encoding as being random variation implementedby the PLM modulator. However if the reserved data sequence is receivedin the control data encoding the receiver will then treat subsequentcontrol data encoding as genuine control data. When the PLM modulatorthen reaches the end of the control data to be transmitted it may send asecond reserved sequence of control data (which may be the same ordifferent to the first reserved sequence) to indicate that the controldata has ended. After the second reserved sequence has been encoded thePLM modulator may then return to varying between the possiblealternative pulse lengths for EMI management reasons. The receiver,after detecting the second reserved sequence will stop treating theencoded control data as genuine control data and disregard the controldata encoding unless and until another instance of the first reservedsequence is detected.

The same principles apply to the embodiments where the timing of boththe rising and falling edges of the data pulse may be varied to encodedata. As mentioned previously if there are eight time slots in atransfer period with the first (or last) time slot always at logic 0 toensure a gap between data pulses then there are 28 possible symbols thatcan be generated (assuming that all combinations of a rising and fallingedges synchronised to a time slot are allowed). This allows four bits ofdata to be encoded per symbol using a set of at least 16 possiblesymbols. This means some possible symbols may be unused—although in someinstance more than one possible symbol may be available for particularencodings with the alternative symbols being selected for DC balance orwhitening the EM spectrum and/or some symbols not used for data encodingmay be used for control or synchronisation purposes. In one embodimenthowever more than 16 symbols may be used, for instance all 28 possiblesymbols could be used, with some symbols also encoding control data andother symbols not encoding any control data. Thus one symbol couldencode a particular data payload value, say 1111, and also encode acontrol data value of 0 and another symbol could encode the same datapayload, e.g. 1111, with a control data value of 1. However othersymbols encoding other data payload values, say 0000 for example, maynot encode any control data.

In some embodiments, especially for higher-speed links, where rise/falltimes may be significant and may be further degraded in transmissionalong the wires, it may be advantageous to ensure that the shortestpulse length that can be used has a certain minimum width. For instancerather than allow the shortest possible pulse length to be equal to1T_(MCLK) it may be beneficial to ensure that the shortest pulse widthis, for example 2T_(MCLK). As shown in FIG. 18 if the frequency of themaster clock signal MCLK is eight times that of the transfer clocksignal TCLK then there are eight time slots and, as shown in plot (a),there may be seven possible positions at which a pulse may end, leadingto seven possible pulse widths. However to ensure satisfactory detectionof the pulse the minimum pulse width may be set to be equal to two timeslots and hence there may be only six possible pulse widths—positions 0to 5 shown in plot (b). There may also be advantages in ensuring aminimum gap between the end of a pulse and the start of the next pulse,e.g. there may be a minimum gap equal to two time slots—either incombination with an extended minimum pulse duration leading to fivepossible pulse widths as shown in plot (c) or instead of an extendedminimum pulse duration, plot (d). It should also be noted that theminimum pulse length and/or minimum gap between pulses could be someother value, such as 1.5 times the second clock period as shown in plot(e).

The PLM modulator described with reference to FIG. 13 could be arrangedto produce such minimum periods by adjusting the values and delay of theramp signal for instance and the receiver would simply arrange the delaytaps at appropriate points and with an appropriate look-up table orequivalent.

Embodiments of the present invention using a fixed rising or fallingedge therefore allow multiple data channels to be transmitted over asingle link with the possibility of alternative encodings beingavailable for at least some combinations of input data, which can beused to reduce the EMI effects by whitening the data signal spectrumand/or allowing transfer of additional data, such a control data,possibly at a reduced data rate.

Bi-Directional Link

The embodiments described have generally discussed transfer of data inone direction e.g. with reference to FIG. 2 from component 201 tocomponent 202. In many applications there may be a need to transfer datain both directions. As discussed previously there may therefore be afirst link for data to be transferred in a first direction, say from afirst device to a second device, and a second link for data to betransferred in the other direction, from the second device to the firstdevice. If the data to be transferred in both directions comprisesmultiple channels then each of the first and second devices may compriseaudio interface circuitry according to the embodiments described above.As also mentioned several devices may be connected together in a chainwith links between successive devices in the chain and data may bepassed around the chain as required.

In some embodiments however the same link may be used for data transferin both directions. FIG. 19 shows one embodiment of bi-directionalinterface circuitry that allows multi-channel data to be transferred inboth directions over the same link 1901 by means of a PLM Codec, i.e. aPLM encoder/decoder. First bi-directional interface circuitry 1902 isarranged to receive a first PLM signal PLM-1 for transmission over thelink 1901 and to extract a second PLM signal PLM-2 that has beenreceived over the link 1901 for onward transmission. Bi-directionalinterface circuitry 1903 does the same but in reverse.

In each of the first and second bi-directional interfaces circuitrydrives switched currents dependent on PLM-1 or PLM-2 onto that end ofthe link 1901. This link has a fixed impedance (at one or both ends) toa reference voltage V_(R) which may, for instance by VDD/2. Both signalsPLM-1 and PLM-2 therefore provide a resultant voltage on the link 1901as illustrated in FIG. 20. The bi-directional interface circuitry isthen arranged to subtract the signal transmitted from that end from theresultant voltage, thus ensuring that the inbound signal derived fromeach bi-directional interface circuit is just that which was transmittedat the far end.

In another embodiment, as illustrated in FIG. 21, a single link 2101could be used as a time-multiplexed bus. Thus with a period of atransfer clock signal TCLK there may be a first period of N clock cyclesof a second, higher rate clock signal MCLK for data transmission in onedirections and a period of M clock cycles of the master clock signalMCLK for transmission of data in the other direction. Thus thetransmitter at one side of the link, say the PLM modulator 2102, maytransmit a pulse-length-modulated data signal based on the input datachannels PDM-1A and PDM-1B where the pulse length is varied between 1and N of the second clock cycles. At the end of the N cycles the outputfrom the PLM modulator 2102 is made high impedance and the PLM modulator2103 at the other end of the link starts transmitting itspulse-length-modulated signal for a selected number of the M remainingcycles. The ratio of N to M may be 50:50 or may depend on the number ofchannels of data being transferred in each direction. The right handside receiver passes data to the data extraction circuitry 2104 onlywhen its transmitter is disabled (set to high impedance). The dataextraction circuitry 2104 recovers the clock signal TCLK and may alsodetermine the master clock signal MCLK, as described earlier, both ofwhich are supplied to the PLM modulator 2103. The pulse length of thereceived data can then be used to determine the output audio data PDM-1Aand PDM-1B. The left hand data extraction circuitry 2105 operates in thesame way except that clock recovery is not needed as the relevant clocksignals are available. The bus keeper or bus holder 2106 operates tokeep the link line 2101 stable (i.e. low) in underlap when both driversare high impedance.

Types of Link

It should be noted that the embodiments described above have describedthe data link as a conductive path. This conductive path could compriseone or more of a conductive path on an integrated circuit, a contactbetween an integrated circuit and a printed circuit board (PCB), aconductive path on a PCB, a contact for an external device such as acontact on a plug or socket for example and/or a conductive wire.

For example an audio signal processing integrated circuit (IC) of a hostdevice, such as an audio codec or audio hub, may have a PLM encoder aspart of the IC for producing a PLM data stream from data received fromthe codec/hub, for instance retrieved from memory or from acommunications processor or the like. This may be communicated, forinstance via a PCB, to output transducers within the host device. Eachoutput transducer may have an associated transducer driver whichincludes a PLM decoder which decodes the data stream for that transduceror one transducer driver may include a PLM decoder for decoding the datastreams for multiple transducers. In some embodiments have the PLMencoder and/or PLM decoder may be separate ICs. There may be a DACassociated with an output transducer to receive, from the PLM decoder,the decoded digital data stream for an output transducer and provideanalogue driving signals. The DACs could form part of the same IC andthe PLM decoder and/or transducer driver or again be a separate IC. ThePLM data stream may also be transmitted to a connection interface forconversion to an analogue signal for transmission to an accessoryapparatus.

The PLM data stream may also be transmitted to an external device, i.e.an accessory device such as a headset, via any suitable connection—forinstance a male plug and female socket. The PLM data stream maytherefore be sent, via suitable contacts, i.e. poles, on the relevantconnectors, to an external peripheral device. Again there may be atransducer driver having an associated PLM decoder and possibly a DAC,as part of the same IC or separate, for each output transducer or onetransducer driver may produce data streams for multiple transducers.

Likewise input transducers may be coupled via ADCs to a PLM encoder forgenerating a PLM data signal to be sent to a PLM decoder associated withthe audio circuitry, which may or may not may part of the same IC.

However the link could be provided by any suitable means of dataexchange. For instance an optical (for example infra-red) data linkcould be provided via a suitable waveguide, such as an optical fibre orthe like, or may simply be transmitted into free space, e.g. for awireless headphone. In such an embodiment the PLM modulator may comprisean optical source for producing an optical pulse with rising and fallingedges at required times during a transfer period and the receiver maycomprise an optical detector. The optical link may comprise a suitableoptical detector and as mentioned the link may comprise opticalwaveguides or be at least partly achieved through free space. Part ofthe optical link may be implemented in an optical circuit board. Theoptical transmitter may itself be a PLM encoder and thus may receiveindividual data streams directly and produce an appropriate PLM datastream. Alternatively the optical transmitter may receive a PLM datastream over a conductive wire from a PLM encoder in a host device andsimply convert it to an optical signal. On a receiver side an opticalreceiver may be an optical PLM decoder for decoding the optical PLM datareceived and providing individual digital data streams for eachtransducer. Alternatively there may be an optical transducer whichconverts an optical PLM data signal into a corresponding electrical PLMdata stream which can then be decoded as described above.

Alternatively or additionally the link could be provided by RFtransmission and thus a PLM modulator may comprise a suitabletransmitting antenna for generating modulated RF transmissions definingdata pulses and the receiver may comprise a receiving antenna. Again theRF transmission apparatus may include a PLM encoder for receiveindividual data streams directly and produce an appropriate RF PLM datastream or it may receive a PLM data stream over a conductive wire from aPLM encoder in a host device and simply produce a corresponding RF PLMsignal. On a receiver side the RF receiver may be an RF PLM decoder fordecoding the RF PLM data received and providing individual digital datastreams for each transducer. Alternatively there may be an RF transducerwhich converts an RF PLM data signal into a corresponding electrical PLMdata stream which can then be decoded as described above.

The link may comprise various different types of transmission medium andmay comprise components for converting one type of PLM data signal, saya voltage signal on a conductive path into another type of PLM datasignal say an optical signal in an optical waveguide for example.

Power Transfer

In some applications, where a destination device for the transmittedaudio signals does not have its own power source it will be necessary totransfer power as well as the data. Thus, although the data for multipleaudio channels may be transmitted via single data link it may benecessary to also have at least a power supply link and a ground link.Where there is two way data exchange there may also be separate datalinks for outbound and inbound data. However in some embodiments,especially when the PLM data signal comprises voltage modulated signals,the data signal may also be used to transfer power. Various techniquesare known for deriving a power supply voltage from a data supply, forexample using a simple diode or synchronous switching circuitry. Whenthe data link is also used to transfer power the voltage on the dataline may be arranged to vary from a first non-zero voltage to a secondnon-zero voltage to indicate the data pulses. In this way the signalline is never at zero voltage. However given that with embodiments ofthe present invention there is always at least one data pulse pertransfer period and there may be a minimum pulse period or controlledaverage pulse period (which may, for example, be equal to half the clockcycle) it would be possible to derive a suitable supply voltage even ifthe data pulses vary between a first non-zero voltage and ground.

Applications—Accessories

Various applications of the invention will now be described. Exceptwhere specifically indicated these applications may be implemented usingany of the embodiments described above and thus may involve transmissionof symbols with a rising and falling edge modulated by the data and alsoto embodiments with a fixed rising or falling edge. Control data may betransmitted and the data may be transmitted in frames as discussedabove. The data link may be implemented as a point to point link or aseries of point-to-point links comprising a linear or chained bus.

FIGS. 22 to 29 illustrate some example applications of the embodimentsof the present invention as applied to peripheral devices such asheadsets, i.e. headphones or earbuds or the like which may or may notinclude microphones for voice data transfer or ambient noisecancellation (ANC). The same principles would also apply to connectionsof other accessory devices. For example speakerphones or the like may beconnected to a host device such as a tablet computer and may havemultiple speakers and/or microphones and may transmit audio data forbeamforming or the like.

FIG. 22 illustrates a two way data exchange between a DSP or codec 2201of a first device 2202, such as a personal media player, gaming deviceor mobile telephone, to a headset 2203 via interface circuitry 2204. Inthis embodiment there is two way data exchange using PLM data signals(PLM_UP and PLM_DOWN). The headset may have interface module 2205 whichreceives the PLM_UP data signal, as well as separate power and groundlinks, and which transmits the PLM_DOWN signal. The device 2202 maytherefore connect to the headset 2203 via a 4-connector plug 2206. Theinterface module 2205 of the headset may receive the PLM_UP signal andtransfer the signal to data extraction circuitry associated with eachloudspeaker 2208, 2207 in a similar fashion as described previously (oralternatively could extract the two PDM data channels and send each dataPDM channel to the appropriate loudspeaker). The interface module 2205may also receive audio data, via individual single channel PLM or PDMdata links (which may use a clock or clocks recovered from the PLM datareceived by the adjacent speaker circuitry), from noise cancellationmicrophones 2209, 2210 associated with each loudspeaker and may producea multi-channel audio signal PLM down to be transmitted back to theDSP/codec 2201.

In some embodiments, as will be described in more detail later, theinterface may be also configured to operate in a legacy mode with olderheadsets which may, for example, only operate with analogue drivingsignals. Thus the interface may include a means (not shown) fordetermine whether the peripheral device is capable of operatingaccording to the present protocol. If not the interface may transmitanalogue driving signals.

Where power is being provided to a peripheral device, especially on anoutput connection that may in legacy mode be used to send or receiveaudio data, there may be current or power limiting applied to the poweroutput until the peripheral device validly identifies itself andrequests more power. This can prevent a relatively large and potentiallydangerous amount of current being supplied incorrectly to a peripheraldevice. Once the peripheral has established its identity it can requestmore power. The initial power supply may therefore be just sufficient toallow the peripheral to identify itself.

FIG. 23 shows a similar arrangement to FIG. 22 but illustrates that theearphones may also comprise error microphones 2301 and 2302 whichmonitor the sound emitted into the ear. The error microphones 2301, 2302may be used instead of outward facing, i.e. ambient noise facing, activenoise cancellation microphones 2209 and 2210 to implement a feedbacknoise cancellation system. In some embodiments however the forwardfacing, i.e. speaker facing, microphones are provided instead or inaddition to the outward facing microphones 2209, 2210 in a combinedfeedback/feedforward system or to continually tune the signal processingof a feedforward ANC system. Where both inward and outward facingmicrophones are present then there may be two channels of audiomicrophone data to be transmitted from the earphone. The errormicrophone may be a digital microphone package or chip converting asignal from the analogue microphone transducer element into a digitalformat, for example a single-bit delta-sigma audio data stream.

Each earphone may therefore be provided with a PLM module 2303, 2304comprising data extraction circuitry, i.e. a PLM decoder, for extractingthe appropriate audio signals for the speakers and also PLM encodingcircuitry for encoding the audio data from the microphones into anPLM_DOWN signal for transmission to the interface module 2205.

In some embodiments the loudspeaker in the earphone may generate asignal indicative of the instantaneous current flow passing through thespeaker coil. This flow data regarding the speaker current may be usefulfor speaker protection functions where the current flow data is fed backto an upstream DSP circuitry to appropriately limit the current fed tothe speaker so as to avoid thermal or mechanical overload of thespeaker. This current flow data may be generated by any suitable currentsensing element, such as a MOS current mirror or resistor, within driveramplifier circuitry inside the headphone. The flow data may be in adigital format, for instance a 1-bit delta-sigma data stream. Thus theremay be a desire to transmit speaker current flow data in addition to oneor more audio data channels from one or more microphones. Againtherefore PLM circuitry in the earphone may combine the microphone dataand current flow data into a PLM signal as described previously fortransmission to the interface module.

FIG. 24 shows an embodiment similar to that shown in FIG. 23 but withaddition of a voice microphone 2401 to pick up a user's voice, possiblytogether with a related outward facing microphone 2402 that picks upambient noise to enable transmission side noise cancelling. Again thesemicrophones 2401 and 2402 may be digital microphones, for example withsingle-bit delta-sigma output formats. The microphones may or not beco-packaged with interface module 2205. The microphone data could betransmitted to the interface module 2205 as, for example, separate PDMdata streams for the interface module to merge this data into a PLMsignal (PLM-DOWN) to be transmitted to the device. However in someembodiments these output audio streams may be merged together by PLMmodulator 2403 which is local to the microphones to produce a PLM datastream for transmission to interface module 2205. The interface modulemay then merge the data from the voice microphone(s) with any data fromthe earphone noise cancellation microphone(s) and any other data such asspeaker current flow data.

FIG. 25 illustrates an embodiment with the same four links as shown inFIG. 22. However in this embodiment the interface module is effectivelylocated with one loudspeaker. Thus the right loudspeaker data extractionmodule (i.e. on the right hand of the page—as shown this would be theleft channel for the user) receives the PLM_UP signal, extracts the datafor the right audio channel and forwards the PLM_UP data onto the leftloudspeaker circuitry. The ambient noise cancellation audio data fromthe microphone associated with the left loudspeaker is sent, as singlechannel PDM, possibly using a clock or clocks recovered from PLM_UP, tointerface circuitry associated with the right loudspeaker where themulti-channel PLM_DOWN signal is formed.

FIG. 26 shows another embodiment similar to FIG. 25 but againillustrates that there may be inward facing and/or outward facingmicrophones and thus there may be data from more than one microphonefrom each earphone. Additionally or alternatively there may be speakercurrent flow data to be transmitted as described above.

In this embodiment the earphone shown on the left thus has a PLM modulefor transmitting a PLM signal based on the combined microphone/currentdata. The interface in the earphone shown on the right combines thisdata with the data from the right earphone for transmission to thedevice 2202. The PLM_DOWN signal from the interface in the right shownearphone may therefore be the combined data from four or more datastreams, e.g. two microphone channels for each earphone.

FIG. 27 shows another embodiment similar to that shown in FIG. 22 butwherein a single link (PLM UP/DOWN) is used for bi-directional datatransfer. Thus only a three connector plug and socket are required. FIG.28 is a similar embodiment to that shown in FIG. 25 but using abi-directional data link. In this embodiment the data link between theright and left hand speakers may be used to send the multichannel PLMsignal up to the left speaker with only one channel of PDM data from themicrophone associated with the left speaker being sent back on the samelink. Again there could be more than one microphone for noisecancellation and/or speaker current flow data could be sensed andtransmitted along the bi-directional link. Also there may be voicemicrophone data from a voice microphone, possibly with a transmissionnoise cancellation microphone.

FIG. 29 shows an embodiment similar to that shown in FIG. 28 but whereinthe data signal link is also used to provide power supply. Thus only atwo connector plug/socket is required to provide the link for power/datasignal PMDM and ground. An interface module in the headset derives powerfor the loudspeakers and data extraction circuitry.

Active Ambient Noise Cancellation

As described with reference to FIGS. 22 to 28 various embodiments of theinvention allow the transfer of audio data between a DSP/audio hub/audiocodec and transducers of an accessory device such a headset for thepurposes of ambient noise cancellation. This represents a particularlyadvantageous aspect of the embodiments of the present invention.

As mentioned above noise cancellation may be of the feedback type, inwhich case microphones monitor the sound transmitted into the ear andadjust the audio data to be transmitted to reduce noise, and/or thefeedfoward type, in which case noise travelling towards the ear isdetected and compensation signals generated. In both types of noisecancellation, but especially for feedforward ANC systems there is a needto generate the noise cancellation data extremely quickly. For afeedforward ANC system the time between the noise wavefront passing theoutward facing microphone (and being detected) and subsequently passingthe earphone/earbud loudspeaker (by which time the noise cancellationsignals should be generated) may be of the order of 40 microseconds orless. Thus it is obviously necessary for ANC, especially feedforwardANC, for any propagation delays in getting the data to and from asuitable processor to be as low as possible.

Generally, ambient noise reduction relates to a speaker housing, such asan ear-bud housing for example, wherein the speaker has an intrinsicresponse time. In other words there is a response time between sendingan audio signal to a speaker and the resulting sound being produced,some of which may be processing delays or the like but some of whichwill be due to inertia of moving parts etc. In use, the loudspeakerdirects sound into an ear of a listener when disposed adjacent an entrylocation to the auditory canal of the ear.

Each speaker housing, for left and right, comprises at least onemicrophone suitably positioned to sense ambient noise approaching theear of a listener and circuitry for converting the sensed ambient noiseinto electrical signals for application to the speaker so as to generatean acoustic signal opposing the ambient noise.

For feedforward-type noise cancellation the time-of-flight of the noisebetween the position at which the noise is detected and the speakerdefines the time available to generate the appropriate noisecancellation signals and driving the loudspeaker taking its intrinsicresponse time into account. As headsets may be relatively small in someembodiments the time-of-flight of the sensed ambient noise from eachmicrophone to the entry location of the auditory canal may besubstantially matched to the intrinsic response time of the speaker suchthat the acoustic signal opposing the ambient noise is generated by thespeaker in substantial time alignment with the arrival of said ambientnoise at said entry location. This clearly requires fast processing ofthe sensed noise.

The acoustic signal and the ambient noise may preferably be time alignedat said entry location to 40 microseconds or less or more preferably 25microseconds or less.

With conventional multi-channel audio data transfer protocols, forexample those relying on using samples of audio signal at conventionalsampling frequencies such as 8 kHz, 16 kHz, 32 kHz, 44.1 kHz, 48 kHz, 96kHz and 192 kHz for example, it is not generally possible to transferthe detected noise data from the outward facing microphones to theDSP/audio hub and suitable cancellation signals back to the loudspeakersin time. Thus conventional noise cancelling headphones/headsets, i.e.accessories, apply the noise cancelling processing within the accessoryitself so as to avoid the inherent delays in transferring data atstandard sampling frequencies to the DSP of the host device, e.g. themobile telephone or computing device. This means that the headset mustdisadvantageously have its own source of power, typically requiring abattery, and on-board processing capability adding to the expense of theheadset.

The data transfer protocol of the embodiments of the present inventionhowever provides a very low latency data link between the microphones ofthe accessory device and the DSP/audio codec. As described previously anoversampled data stream, such as a 1-bit PDM data stream, can betransferred at a bit rate of the order of 3 MHz and multiple channels ofdata can be transferred per transfer period. This provides a very lowlatency data link meaning no significant time is wasted in transferringdata from the outward facing microphone or transferring the noisecorrected audio data to the loudspeakers of the microphone. The point tomulti-point connection can avoid any latency on a bus but even in achained arrangement the latency introduced by each component in thechain may be as low as single transfer period. Thus embodiments of thepresent invention actually enable noise cancellation processing for anaccessory to be performed by the DSP/audio hub/codec of a host device.This represents a novel and advantageous aspect of embodiments of thepresent invention.

Also as illustrated in FIG. 25, all required channels of audio data maybe transmitted up and down through respective single poles or pins on acommonly used 3.5 mm four-pole jack plug-socket connection, or similar(the other two poles/pins being used for supply and ground). Thiscontrasts with analogue or non-multiplexed systems where extra poles areneeded, leading to either a bulky or a fragile and less electricallyreliable connector.

Applications—Device

As mentioned some embodiments may include transmission of detectedcurrent and/or voltage waveforms from the speakers to the host device orcodec, for speaker protection features, instead of or in addition tosignals from the adjacent microphones. Also similar configurations ofthe speakers and microphones may be part of the host device, wired via aPCB or other internal wiring, rather than separately wired headphones.

A device may therefore have several different audio transducers such asspeakers and/or microphones within a device and/or connections forvarious external audio peripherals such as headsets etc.

FIG. 29 illustrates how a device 2901, which could for example be aportable computing device, mobile telephone handset or the like, mayhave an audio codec 2902 that may or may not include Digital SignalProcessing (DSP) circuitry for processing audio signals for the device.The device itself may comprise various audio transducers, i.e.components such as loudspeakers 2903 and 2904 and microphones 2905 and2906. In addition there may be a connection interface 2907, such as asuitable jack, for connecting a headset 2908 which may compriseloudspeakers 2909 and 2910 for audio play back and one or moremicrophones 2911 for voice communication or noise cancellation. It willof course be appreciated that the headset could take the form of any ofthe embodiments described above.

In operation the DSP/Codec 2902 may transmit data to the loudspeakers2903 and 2904 of the device or the loudspeakers 2909 and 2910 dependingon whether a headset is connected and/or the operating mode of thedevice. The DSP/Codec may also receive data from the microphones 2905and 2906 on the device and/or microphone 2911 on the headset. There mayalso be data regarding operation of the transducers such as loudspeakercurrent data as described previously.

It can be therefore be seen that, in the audio domain, there device hasa number of possible generators and consumers (or sources and sinks) ofdata that need to communicate with the DSP/Codec. For instance,microphones 2905 and 2906 on the device may be generators of audio datastreams, labelled as B and C respectively. Loudspeakers 2903 and 2904 onthe device may be consumers for audio data streams (labelled as streamsa and d respectively) transmitted from the DSP/Codec 2902. In additionin some embodiments the loudspeakers may also be generators of currentdata, data streams A and D respectively.

The loudspeakers 2909, 2910 of the headset may also be consumers ofaudio data streams (streams e and f respectively) and headset microphone2911 may be a generator of data stream G.

All of the signals generated and/or consumed by the audio transducersmay be digital signals, for instance 1 bit data streams, with anynecessary digital-to-analogue or analogue-to-digital conversion beingembedded within the respective transducer circuitry.

All of the transducers within the device 2901 and interface 2907therefore need to be connected to the DSP/Codec 2902, and, as mentionedabove, digital data connections are typically preferred. Each transducerwithin the device could be separately connected to the DSP/Codec 2902via a separate connection but this would involve multiple connectionpaths on the PCB and multiple pins on the DSP/Codec.

As mentioned previously the use of PLM encoding can allow multiple audiodata streams to be encoded and transited over a single wire link. Thusas discussed above in relation to FIG. 2 the data streams a and d forloudspeakers 2903 and 2904 could be combined into a PLM signal andtransmitted over a common signal path, i.e. a single signal path on thePCB say, for at least part of the signal path before being separatedinto separate paths to each loudspeaker.

In another embodiment however the signal paths may be arranged so thatthere is a link, i.e. signal path, between first and second componentssuch that data to be transmitted to or received from the secondcomponent is transmitted via the first component.

In other words for data to be transmitted to the second component, a PLMsignal with combined data for both the first and second components maybe received by the first component. The data extraction circuitry forthe first component can extract the relevant data for the firstcomponent and then forward the signal to the second component. Theforwarded signal may be an unchanged version of the received PLM signaland the second component may therefore extract the relevant data.Alternatively the first component could extract the data for both of thefirst and second components and transmit a PDM signal with just therelevant data for the second component to the second component. In someembodiments, as will be described below, the second component may itselfhave a link for forwarding a signal to a third component and the datatransmitted to the first component may be a combined data stream for allof the first, second and third components, in which case the dataforwarded from the first component to the second component may be theoriginal PLM signal or a re-encoded PLM signal with just the data forthe second and third components.

Likewise for data being received the second component may transfer datato a first component over a link—which could be a single data streamrepresenting data from just the second component or may be a PLM signalencoding data from at least a third component. The first component willreceive the data and merge it into a PLM signal for onward transmission.

FIG. 30 shows an embodiment of a suitable connection between thetransducers of the device arrangement illustrated in FIG. 29. FIG. 30shows a PLM module 3001, e.g. a PLM codec, for communications to/fromthe DSP/Codec 2902. The arrangement shown in FIG. 30 may make use of thebus protocol as described previously.

For data to be transferred from the DSP/Codec 2901 to the loudspeakersthere is a first signal path to the loudspeaker 2903 for transmitting asignal m0. There is also a signal path from loudspeaker 2903 toloudspeaker 2904 for transmitting a signal m1 and a further signal pathfrom loudspeaker 2904 to interface 2907 for transmitting a signal m4. Inuse, with a headset connected by plugging a suitable connector into thejack of interface 2907 there will also be a signal path from interface2907 to the loudspeakers of the headset 2908 for transmitting a signalm4′.

The signal m0 transmitted by the DSP/codec 29202 to loudspeaker 2903must therefore contain all necessary audio streams for loudspeakers2903, 2904 and the loudspeakers of the headset 2908, i.e. signal m0 mustcomprise audio streams a and d and/or e and f.

In some devices and applications the device may be arranged such thatthe internal loudspeakers 2903 and 2904 do not operate if audio is beingplayed via the headset loudspeakers and vice versa. In such a situationthere may only ever be a need to transmit two audio data streams,perhaps along with control data for activating/deactivating the relevantloudspeakers. In some applications however it may be necessary totransfer separate data streams a, d, e and f simultaneously, possiblywith control data for controlling the various audio components.

Signal m0 may therefore be a PLM signal which encodes the required datastreams. As mentioned previously using a master clock which has afrequency which is eight times that of the clock used to define thetransfer period four bits of data may be encoded per symbol. If controldata is also sent two of the required audio data streams may be sent byinterleaving the bits for said data streams.

Data extraction circuitry associated with loudspeaker 2903 extracts thedata stream a for loudspeaker 2903 (if present) and forwards the signalto loudspeaker 2904. The signal m1 which is forwarded may be are-encoded PLM signal with just data streams d, e and f but it may besimpler in some embodiments to simply forward the signal received, i.e.m1 is equivalent to m0 (subject to any modifications of any controldata).

Data extraction circuitry associated with loudspeaker 2904 receives thesignal m1 and extracts any data for audio stream d. A signal m4 is thenforwarded to the interface 2907. Again the signal that is forwarded maythe same as the signal received, i.e. m4=m1 or the signal may bere-encoded.

The interface 2907 receives signal m4 and forwards data to the headset(when connected). The interface jack may also transmit power and providea ground connection as described above in relation to the headsetembodiments. The interface may simply provide an appropriate connectionfor the incoming signal m4 to the jack socket. However in someembodiments it may be preferred to receive the signal at the interfaceand re-transmit a signal via the connection. Again the transmittedsignal m4′ may be the same as the signal m4 received at the interface.However in some embodiments the interface does extract data streams eand f and generates a new PLM signal containing only data streams e andf. As will be appreciated the headset connection may be relatively longand of a more variable quality than is the case for the relativelypredictable connections within the device itself. Re-encoding the signalto be transmitted by the interface to contain just the audio datastreams e and f for the headset means that fewer symbols are required inthe signal and thus a larger time resolution (i.e. slower second clock)can be used as compared to the signals transmitted within the device.This can ease the signal integrity requirements and ensure goodtransmission.

The discussion above of course assumes that the headset is suitable forreceiving PLM signals. In some embodiments the headset may instead useconventional digital audio signals or instead require analogue drivingsignals. In which case the interface 2907 receives the PLM signal m4 andextracts the relevant audio data streams e and f and generatesappropriate driving signals for the headset.

FIG. 30 also shows a separate series of signal paths between componentsfor transmitting data to the DSP/Codec 2902. Data from the microphone(s)of the headset 2908 may be communicated to the interface 2907 as a datasignal M4′. Where there is a single microphone with digital output thisoutput could be transmitted as a single-bit digital data stream. Wherethere are two or more microphones and/or other data to be transmitted asuitable PLM signal may be transmitted.

The interface 2907 receives this signal and forwards a signal M4 to nextcomponent in the series, in this example loudspeaker 2904. Dataextraction circuitry in the interface 2907 could determine the receivedaudio signal data G in the received signal M4′ and re-encode it in thesignal M4 transmitted to the next component or the received signal maybe transmitted directly in some embodiments.

In this example the loudspeaker 2904 is also a generator of data, forexample current data used for speaker protection. Thus the signal M4from the interface is received and combined with the speaker data foronward transmission as a PLM signal M3 to the next component, microphone2905. Here the data from microphone 2905 is combined with the existingaudio data and a new signal M2 transmitted onward.

Data stream C from microphone 2906 is also combined into the signal toform a new signal M1 which is then combined with data A from loudspeaker2903 to form a signal M0 which is then passed to the PLM module 3001associated with the DSP/Codec 2902. Signal M0 is therefore a PLM signalwhich includes data streams A, B, C, D and G. The data extractioncircuitry in PLM module 3001 can extract the individual data streams andtransmit the retrieved data streams to the DSP/codec in any suitableway.

It should be noted that each of signals M0, M1, M2 and M3 are PLMsignals. As mentioned signal M4 comprise a single audio data stream onlyand thus could be a PDM signal for instance but in some embodiments maybe a PLM signal.

In each link between components the PLM signal produced may be providedwith just sufficient time slots to encode the audio data streamspresent. For example if there is only a single data stream of audio datafrom the headset 2908 and a single data stream of current data fromloudspeaker 2904 then signal M3 may be an PLM signal with four timeslots to provide six possible different symbols—of which four may beused to encode the data. (Alternatively in an embodiment with fixedrising edges say there may be five time slots to provide four differentpulse lengths with a gap between pulses. In other words the second clockfor the relevant link may have a frequency five times that of thetransfer clock signal TCLK). For signal M2 however there is a need toencode another data stream (data stream B) and thus eight differentpossible symbols and at least five time slots may be required (nine ifcoded with a fixed edge). Thus the relevant link may run with adifferent second clock frequency which is five (or nine) times the firstclock frequency. Typically the same first clock frequency will be usedon all links as this will be set by the required bit rate.

In some embodiments however it may be preferable to use the same generalcircuitry for each link, thus the first and second clock frequencies maybe set to be the same for each of the PLM signals in the chain. Also itmay be desirable to encode control data as described previously. Thusthe bus arrangement as described previously may be used with signal M0may have a master clock frequency sufficiently fast to encode the fivedata streams A, B, C, D and G together with control data (with all datastreams encoded by a separate channel or with at least some of the datastreams being interleaved on a channel of the PLM data link).

Each of the components receiving a PLM signal and adding new data to thesignal sent onwards could be provided with data extraction circuitry forextracting the existing data and a PLM modulator for producing a new PLMsignal based on an appropriate truth table.

In embodiments with a fixed rising or falling edges it is noted that insome instances a component may be arranged to modify the pulse length ofthe data pulse received without determining what the existing data is.For example assume that each of signals M4, M3, M2, M1 and M0 is a PLMdata signal with 32 available pulse lengths, i.e. there are 33 timeslots in each period of the transfer clock signal TCLK. The interface2907 may produce a signal that encodes the single data stream G only.The PLM associated with the interface could be arranged to produce apulse of 1 time slot in length for data 0 and a pulse of 17 time slotsin length for data 1. The loudspeaker may then may arranged to leave thepulse length unmodulated if data stream D is data 0 but to extend thepulse length by 8 time slots if data stream D is data 1. Likewisemicrophone 2905, microphone 2906 and loudspeaker 2903 could each bearranged to leave the pulse length unmodulated if the relevant datastream is data 0 and to extend the pulse length by 4, 2 and 1 time slotsrespectively if the relevant data stream is data 1.

The result will be a pulse length in signal M0 which uniquely encodesall of the data streams A, B, C, D and G. However each component in thechain simply modifies the pulse length by a predetermined amount basedon its own data stream and doesn't need to determine what the upstreamdata is.

This arrangement means that a single PLM signal can be sent upstreamfrom the DSP/Codec to provide data for all consumers of audio data witheach consumer component simply tapping data from the signal at anappropriate point in a chain. Likewise a chain for downstream data to betransmitted to the DSP/Codec is provided with data being merged into thesignal at appropriate points. This avoids the need for distinctconnections to/from each component. The bus protocols described abovemay be used in such an embodiments.

In such an arrangement the upstream and downstream links may be arrangedrelatively close together as shown and thus two signal paths, e.g.traces on a PCB, may pass near to some of the transducers. In theexample shown in FIG. 31 the signal paths are close to the transducersconsuming signals a and d and generating signals A, B, C and D. Howeversome transducers may be separated relatively far apart from one anotheron the device. In the example shown in FIG. 31 microphone 3006 islocated on the opposite side of the device. Thus means that a separate,relatively long, connection must be used to connect the microphone to asuitable tap point on the PLM link. Otherwise a separate connectionbetween microphone 3006 and DSP/Codec 3002 may be provided.

In another embodiment however the links between at least some componentsmay be arranged in a complete chain. Thus data transmitted by theDSP/codec 3002 for an upstream consumer component may be transmitted inthe same direction as data generated by a component for transmission tothe DSP/Codec 3002. This allows for two signal paths to be spaced apartin a configuration that reduces the amount of signal track required butstill provides communication between the DSP/Codec 3002 and eachconsumer and still allows each generator to transmit to the DSP/Codec.

FIG. 31 illustrates one example of a suitable arrangement. In thisarrangement the PLM module 3001 of DSP/Codec 2902 has a first link toloudspeaker 2903 as described previous and transmits a signal m0 whichis a PLM signal which encodes data for all of the generators asdescribed previously. In this embodiment however the data stream a forloudspeaker 2903 is extracted and the measured current flow data is alsoencoded into the signal transmitted m1 onward from the loudspeaker. Thissignal is transmitted to the microphone 2905 where the signal isre-coded to include data stream B. Signal m2, which thus contains atleast audio data streams d, e and f and data streams A and B, is thenpassed to loudspeaker 2904 where audio data d is extracted for theloudspeaker and current flow data D is encoded. The signal comprisingdata streams e and f and also A, B and D is then passed to the interfacewhere audio data streams e and f can be extracted for sending to theheadset as described above. Any microphone data from the headset canencoded into the signal m4 which is then sent, via a different returnpath, to microphone 2906. Microphone data stream C is encoded to producesignal m5 which is then returned to the DSP/Codec which can the extractdata streams A, B, C, D, and G.

The PLM signal transmitted between any components may therefore besuitable to encode each of the data streams a, d, e and f and A, B, C, Dand G. It will be appreciated however that no signal comprises audiostreams a and A. In other words as loudspeaker 2903 consumes audiostream a but generates data stream A the PLM signal m1 may be adjustedso that the encoding for outgoing audio stream a is replaced by theencoding for measured data stream A. Likewise no signal contains bothaudio stream d and data stream D or audio streams e and f and microphonedata stream G. The most number of individual data streams is thus thosecontained in signals m2, m3 and m5 which need each need to encode fivedifferent data streams (m2 encodes consumer audio streams e and f andgenerated data streams A, B and D; m3 encodes data streams e, f and A, Band D; and m5 encodes generated data streams A, B, C, D and G).

Thus each link may use a PLM signal capable of encoding five differentdata streams. Signal m0 originally transmitted will encode the consumerdata streams a, d, e and f and may use the other available symbols asalternatives to whiten the data spectrum as discussed previously. Signalm1 re-encodes the signal to remove the data encoding for stream a but toinclude data from stream A.

Again the PLM module associated with each component may extract the datafor each data stream and remodulate taking any new data into account.

Obviously the data extraction/modulation circuitry for each componentwill need to know which data stream to extract. This can bepredetermined and fixed so that each transducer always extracts a givendata stream, in which case the relevant details can be stored in asuitable non-volatile memory and/or the data extraction circuitry may behardwired so as to generate data 1 for predetermined symbols and data 0for other symbols. In other embodiments however there may be someconfigurability in which component extract data at which point in thechain. This could be established however by any suitable configuration,such as token passing such as is known in the art.

Compatibility with Legacy Systems

As described above the data interfaces of the present invention may beused in communication between various electronic devices and accessorydevices such as headsets etc. Typically devices such as mobiletelephones may be operable with a set of headphones and thus such adevice will have a connector, usually a suitable socket, for connectingto a plug associated with the set of headphones and via which audio datamay be transferred.

Conventional headsets may have a connector, for example the common 3.5mm three pin TRS jack plug for the transfer of analogue audio data. Twoof the pins are arranged to receive left and right analogue audiochannels respectively, and the third pin is a ground pin. Some headsetsmay also comprise a microphone, for instance for voice communication,and such headsets may typically have a four pin connector, such as a 3.5mm TRRS jack, with the additional pin being for communicating ananalogue microphone signal. Often such headsets may not have their ownsource of power and thus the analogue audio signals received are used todrive the loudspeakers directly and the microphone may not require aspecific power source (typically being biased from a fixed voltage viaan off-chip resistor of about 2.2 kΩ and providing a signal bymodulating this resistive bias). This analogue audio-out connection canalso be used to connect the device to other audio accessory orperipheral devices and may drive a line-level output when required.

Whilst digital data transfer may be preferred many users acquiring adevice such as new mobile telephone or the like may have legacy headsetsor other peripheral devices that they would like to use with the device.Likewise a user acquiring a headset which is capable of digital datatransfer may nevertheless wish to use such a headset with a device thatonly outputs analogue audio signals. It would therefore be advantageousfor the data interface to be able to operate in a legacy analogue modeas well as in a digital mode according to the protocols describedpreviously.

FIG. 32 shows an embodiment of a suitable interface module that allowsaccessory apparatus to use standard connectors for digital data transferand also to function with legacy devices outputting, and possiblyreceiving, analogue audio signals.

FIG. 32 shows a jack plug connector 3201 which, in this embodiment hasfour pins or contacts, 3201 a-d, such as a TRRS type plug.

One pin, 3201 d, of jack 3201 provides a ground connection and isconnected to an appropriate ground path of the accessory apparatus. Theother three pins 3201 a-c are connected, via signal paths HP LEFT, HPRIGHT and MIC to series switches 3202 a-c respectively which connect toloudspeakers 3203 and 3204 and microphone 3205 respectively.

Series switches 3202 a-c can thus connect the pins 201 a-c of the jack3201 directly to the transducers 3203, 3204 and 3205. Thus in a legacymode of operation where analogue audio signals are used, the seriesswitches 3202 a and 3202 b can connect pins 3201 a and 3201 b of thejack directly to the left and right loudspeakers 3203 and 3204respectively so that the received analogues audio signals can drive theloudspeakers. The microphone may be resistively biased through the MICline and, in use, superimposes a small signal modulation that istransmitted via pin 3201 c to the device to which the accessory isconnected enabling read-out of the microphone data. The microphoneground reference and speaker coil ground current return use the groundconnection provided by pin 3201 d.

In some embodiments the accessory may not have its own source of power.The series switches 3202 a-c are therefore configured to default toproviding a connection between the analogue signal paths and thetransducers. Suitable switches include normally-closed relays ordepletion mode FETs with gates grounded for example via a resistor.

In addition the three pins 3201 a-c are also connected to paths DATA_UP,DATADOWN and PWR to digital processing circuitry 3206. The paths frompins 3201 a and 3201 b, which carry the analogue driving signal for theloudspeakers in legacy mode, default to high impedance.

In a digital mode of operation pin 3201 c is used to supply power to theheadset. When power is available on the PWR/Mic line the digitalprocessing circuitry 3206 may operate to switch the series switches 3202a-c to disconnect the signal lines HP LEFT, HP RIGHT and MIC from thejack pins and to connect the transducers to the DACs 3207 and 3208 andADC 3209.

The digital processing circuitry comprises a discrimination oridentification circuit 3210 which determines whether the device iscapable of using the appropriate digital protocol.

There are a number of ways in which the discrimination circuit canoperate. For instance a device could be arranged, on detection ofinsertion of a jack into a socket (either during operation or on devicestart-up/wake-up/reset etc.) to provide power via pin 3201 c and attempta brief digital handshaking sequence with the connected accessory, forexample by transmitting a first data sequence to the accessory via pin3201 a and monitoring for a second data sequence (which may be the sameor different to the first data sequence) received via pin 3201 b. Onreceiving power the digital processing circuitry of the accessory maytransmit the second data sequence to the device, possibly in response toreceipt of the first data sequence. If the handshaking is successful thedevice and accessory will then transmit and receive digital data. If thehandshaking is not successful both device and accessory may revert tolegacy mode using analogue communications. For example if the accessoryis connected to a device in which pin 3201 c appears to provide suitablepower but the device does transmit the first data sequence the accessorymay continue to operate in legacy mode. Other types of discriminationcircuitry may be used however. For example the signals received via anyof the pins 3201 a-c may be analysed when suitable power is availablevia pin 3201 c to determine whether the characteristics match those ofthe expected digital signals or analogue signals or to look forparticular modulations which may be embedded in the signals.

Assuming that the discrimination circuitry 3210 determines that thedevice is capable of using the appropriate digital protocol a controlsignal may be generated to control the switches 3201 a-c as describedabove. It will be appreciated that the switches 3201 a-c may comprisemore than one switch element for disconnecting the transducers fromsignal lines HP LEFT, HP RIGHT and MIC and connecting them to the DACs3207 and 3208 and ADC 3209.

PLM processing circuitry 3211 will also be enabled and data linesDATA_IP and/or DATA_DOWN may be switched out of high impedance mode asrequired. Power received via line PWR may also be distributed to othercomponents within the accessory that require power in digital mode.

In digital mode data may be received via pin 3201 a and supplied to thePLM circuitry 3211. Data extraction circuitry, i.e. a PLM decoder, canextract the left and right audio data streams as described previously.The relevant signals can then be supplied to DACs 3207 and 3208associated with loudspeakers 3203 and 3204 respectively, for example as1 bit PDM signals, to generate the analogue driving signals for thespeakers.

Signals received from microphone 3205 will be converted to digital byADC 3209, for instance a 1 bit PDM data stream and passed to PLMprocessing circuitry. As described previously if there is only onechannel of audio data to be transmitted to the device then a PLM signalencoding only one 1-bit PDM data channel may be transmitted.

However the accessory may have one or more additional microphones 3212and 3213 for noise cancellation. In legacy mode these microphones maynot be operational and no noise cancelling is applied. However indigital mode these microphones may be operational and the signals fromthe microphones may be digitised and received by the PLM circuitry 3211.The PLM circuitry may therefore generate a PLM signal encoding all audiodata channels to be transmitted to the device as an PLM_DOWN signal viapath DATA_DOWN and pin 3201 b.

It will therefore be seen that the interface embodiment shown in FIG. 32is able to operate in both an analogue legacy mode with devices thatcommunicate using analogue signals and also in a digital mode withdevices able to communicate using digital signals. The interface can usea standard connector suitable for legacy devices. In digital modeadditional functionality may be enabled due to the increased possibilityof communication. In the event that the accessory is plugged into adevice unable to support digital communications it will default tolegacy mode operation.

The embodiment described above with relation to FIG. 32 relates to anaccessory which, in legacy mode, receives two audio data streams andprovides a voice microphone data stream, each via separate pins of afour pin connector (the fourth pin being ground). Some legacy devicesmay however use a three pin connection, for example a music player maybe provided with a three pin socket, i.e. a socket with three separatecontacts, for providing left and right audio data and ground.Embodiments of the invention could therefore be implemented with a threepin connector, for example with one pin used for receiving digital PLMdata, one pin for receiving power and the third pin for ground.

In some embodiments however the interface circuitry may be arranged toderive power by using the received data signal to provide power throughvarious known power harvesting techniques. In one known technique thereceived data signal line, which will swap between a high voltage leveland a low voltage level according to the data, may be connected via asuitable diode to a capacitor. In use when the voltage level is high andthe capacitor is relatively uncharged the diode will forward bias andcharge the capacitor which can then be used to supply power for the PLMprocessing circuitry. This will result in a voltage drop on the signalline but the sending device may be configured to use high and lowvoltage signal levels which are suitable for providing power and alsofor ensuring good signal quality. Other power harvesting techniques areknown and could be used as appropriate. A PLM signal is particularlysuitable for use in such power harvesting techniques as there is a datapulse, and hence high voltage, every transfer period.

The use of power harvesting in this way could reduce the number of pinsrequired and/or allow additional functionality for a given number ofpins. For instance a headset could be provided with a three pinconnector. In legacy mode the three pins may be used for left audio,right audio and ground. Thus the headset simply receives audio data. Indigital mode however one pin may be used for receiving a digital datasignal and, as described above, power harvesting may be applied to thissignal line to provide power for the headset. A second pin couldtherefore be used to send data from the headset to the device, forinstance audio data from ANC microphones or a voice microphone orcontrol data, thus enabling various controls on the headset. The thirdpin could be ground. Thus an accessory with a three pin connector couldbe connected to some devices, such as an older music player, to operatein legacy mode and simply receive analogue stereo audio. The sameaccessory could also be used with a different device however to receivedigital stereo audio and also to provide signals for voice data, noisecancellation and/or headset based control—all via a connector with justthree contacts.

It will also be appreciated that some devices may use different pins fordifferent functions. For instance with the standard TRRS connector foranalogue audio signals there is a convention regarding which pins areused for left and right audio, but some devices use an alternativearrangement for the mic and ground pins compared to other devices.

In some embodiments the interface circuitry may be able to determinewhich pin-out configuration the connected device is using, e.g. withrespect to Mic and ground, and may be able to swap the coupling of pinsof the plug 3201 to paths of the interface circuitry accordingly. Thusif the accessory were connected in use to a device that used therelevant contact for pin 3201 c for ground and the relevant contact forpin 3201 d for microphone signals, this could be detected. The couplingbetween pins 3201 c and 3201 d and the respective paths shown in FIG. 32could then be swapped. Various techniques are known for detecting thepin-out configuration of the device (in terms of use of Mic and Groundpin out configurations), for instance by detecting the resistive biasingapplied to the contact which is used for the microphone and variousmeans are known for automatically swapping the coupling to the signalspaths, any of which could be applied to embodiments of the presentinvention.

It will be appreciated that in the embodiment shown in FIG. 32 one pin(pin 3201 b) which is used for receiving analogue signals in the legacymode is actually used for outputting digital data in the digital mode.It will of course be appreciated that in other embodiments the roles ofpins 3201 a and b in the digital mode could easily be swapped.

FIG. 33 shows another embodiment of an accessory device interface. Thisembodiment has several of the same components discussed above inrelation to FIG. 32. In the embodiment of FIG. 33 however the signalpaths HP LEFT and HP RIGHT includes ADCs 3301 and 3302 respectively forconverting the received analogue signals into digital signals. Thisallows various digital signal processing techniques to be applied withinthe accessory even when the legacy device transmits analogue signals.There may therefore be a digital signal processor (DSP) 3304 arranged toreceive digital signals from the ADCs 3301 and 3302 and possibly noisecancellation microphones 3305 and 3306, for example to perform noisecancellation or spectral equalising to compensate for speakershortcomings etc.

The digital signals can then be converted into analogue signals by DACs3307 and 3308 associated with the loudspeakers.

Likewise in legacy mode the signals from voice microphone 3205 may bedigitised in ADC 3309, optionally subject to any digital processing, andthen converted back to an analogue signal in DAC 3303 to be transmittedover the Mic line to the connected device.

In digital mode, as described previously, the Mic line may be used tosupply power. In this embodiment however the serial switches may belocated within the digital domain of the circuit and thus connect theoutput of the PLM circuitry with the inputs to DACs 3307 and 608 andoutput of ADC 3309. In some embodiments DSP 3310 may be arranged toprocess the signals output from and input to the PLM circuitry, possiblyin addition to as an alternative to DSP 3304.

This embodiment does require at least some power available in legacymode however this could be provided by a battery in the accessory whichmay be recharged, for example when operating in digital mode. In somearrangements, if the power requirements of the digital processing isrelatively low, it may be possible to derive the power needed from theaudio signals themselves using similar power harvesting techniques tothose described above but based on the received analogue signals.

Like the interface embodiment shown in FIG. 32, the embodiment shown inFIG. 33 is also able to operate in both an analogue legacy mode withdevices that communicate using analogue signals and also in a digitalmode with devices able to communicate using digital signals, again usinga standard connector suitable for legacy devices.

It will be noted that both of the embodiments shown in FIGS. 32 and 33are operable in a digital mode that uses pulse length modulation toencode one or more streams of digital data. The PLM digital signal mayalso be used to encode other data streams as well. For instance controldata could be encoded into the pulse length to allow the device toprovide control instructions for the transducers and/or any DSP of theaccessory. Likewise control data such as indicating button presses etc.may be transmitted from the headset to the device in digital mode.Additional data from the accessory device, such as data on the currentflowing through a loudspeaker coil may also be measured and transmittedto the device for speaker protection for example.

In some embodiments the accessory may be operable in more than onedigital mode of operation. The use of the PLM digital format asdescribed is advantageous due to the fact that high data rates can beachieved with continuous communication in one direction over a singlewire without requiring transmission of a bit clock or the like and withgood fidelity. However other digital formats could be supported.

For instance, in a second digital mode pins 3201 a and 3201 b may beused together for differential digital signalling or a 1-bit PDM signalmay be received on pin 3201 a with a bit clock signal being received onpin 3201 b, again with pin 3201 c being used for power and pin 3201 dbeing used for ground. In both of these modes full duplex communicationis not possible but this would be suitable for receiving data only, forinstance just for audio playback or two-way half-duplex communicationcould be established which may be suitable for some applications.

The discrimination circuitry of the digital processing circuitry maythus be configured to not only identify whether digital communication issupported but also the format of digital communication.

The embodiments have been described in relation to transfer of multiplechannels of audio data but it will be appreciated that the principlesmay be applied to other types of data for driving transducers such ashaptic transducers, ultrasonic transducers, hearing aid coils and thelike.

FIG. 34 shows an embodiment of interface circuitry that may be used inan electronic device to communicate with an accessory.

FIG. 34 illustrates a device 3401 which includes an interface 3402, forinstance a codec, for communicating with an external accessory apparatuswhen connected via connector 3403, which in this example may be a socketconfigured to receive a standard connector such as a TRRS jack plug. Inthis example therefore the connector has four separate pins or contactswhich each are coupled to signal lines of the codec 3402.

The interface is capable of operating in a legacy mode for transmittingand receiving analogue signals and also in a digital mode fortransmitting and receiving digital signals.

The PLM codec 3402 may include a digital signal processor (DSP) 3404 forcommunicating with other parts of the host device. For example, the DSP3404 may be arranged to communicate via a first interface 3405 with afirst device component, such as an applications processor 3406 of thehost device. The DSP may also communicate via a second interface 3407with a second device component, such as a wireless codec (e.g. used forwireless and/or RF baseband communications). In some applications theDSP 3404 of codec 3402 may also communicate with further device systems,such as via interface 3409 with a Bluetooth® codec 3410. Note that aswell as providing a communications interface between any of theapplications processor 3406, wireless codec 3408 and BT codec 3410 andthe attached accessory device, the codec may provide a path forcommunications between at least two of the device components, preferablya digital only path.

In operation in legacy mode audio data to be transmitted to theaccessory device may be received by DSP 3404 as digital data and anynecessary processing applied. Individual left and right audio streamswill then be transmitted to digital to analogue converters (DACs) 3411and 3412 respectively. The output from DACs 3411 and 3412 are thenpassed, via HP Left and HP Right signal paths respectively, to therelevant pins/contacts of connector 3403, possibly via suitableamplifiers 3413. In legacy mode, signals received on the mic signal pathfrom the relevant pin/contact of connector pass to switch 3414 whichwill connect to the input to ADC 3415. ADC 3415 will convert anyreceived microphone signals into a suitable digital signal which is thenpassed to DSP 3404, possibly for communication onward to wireless codec3408 say.

In digital mode the switch 3414 may be switched to connect the path usedfor MIC in the legacy mode to a supply voltage VB/VDD Amplifiers 3413may also be disabled and the DSP 3404 will transmit any data to betransmitted to PLM module 3416. PLM module 3416 will encode the datainto a PLM data stream to be transmitted to the device on the DATA_UPsignal path as described previously. Digital data may also be receivedfrom the device on the DATA_DOWN signal path and decoded intoappropriate data streams and transmitted to DSP 3404.

Operation in digital or legacy mode may be controlled by discriminationcircuitry 3417. This circuitry may, on detection that an accessoryapparatus is connected—either during operation or at power-on orreset—determine whether the accessory apparatus is capable ofcommunicating using the relevant digital protocol. For example thedevice may initially start in digital mode and attempt a handshakingexercise as described previously. If the handshaking is successful thecodec 3402 may continue in digital mode. However if the handshaking isnot successful then the codec 3402 may revert to legacy mode and makethe Data_up and Data_down lines high impedance and generate a controlsignal to enable amplifiers 3413 and switch 3414 to connect the Mic lineto ADC 3415.

The codec 3402 is thus capable of communicating with an accessory devicethat supports PLM digital communications, such as described above inrelation to FIGS. 4 and 5, but also with legacy accessory devices usinganalogue signals.

Host Devices

A PLM encoder/decoder such as described above may be implemented as partof a host device which is configured for communication with variousother components of the host device and external devices that may beconnected in various ways. For instance FIG. 35 shows a device 3501, inthis example a mobile telephone handset which may have various internalaudio transducers and connections for external accessory apparatus. Thehandset 3501 may have at least one internal loudspeaker 3502 and atleast one microphone 3503. There may also be various microphones 3504arranged for noise cancellation. There may also be a user interface,such as control buttons 3505, a keyboard and/or touch screen 3506 foractivating various control functions such as volume control.

As shown in FIG. 36 a device 3501 may therefore comprise audioprocessing circuitry such as a digital audio hub 3601 for control overaudio for the device. The audio control circuitry 3601 may thereforeprovide audio data to output transducers of the host device 3501, suchas speaker 3502, and receive data from input transducers of the hostdevice, such as microphone 3503 (and/or microphones 3504 shown in FIG.35).

In use the audio circuitry 3601 may receive and transmit audio datafrom/to various other device systems. For instance the audio circuitry3601 may receive audio data from a baseband (communications) processor3602 via a first interface 3603. The baseband processor 3602 may providecommunications to a telephone network 3604 via a suitable antenna. Thusaudio circuitry 3601 may receive audio data from the baseband processorfor processing and transmission to loudspeaker 3502 and data frommicrophone 3503 may be sent for communication via the telephone network3604.

The audio circuitry may also communicate with an applications processor3605 via a second interface 3606. The applications processor mayretrieve audio data from memory 3607, for instance for playback ofstored music. Commands from a device user interface may be communicatedto the audio hub 3601 via the applications processor 3605.

There may also be additional systems such as a wireless transceiver 3608which communicates via interface 3609 for wireless communications.

In some instances the various device systems may communicate audio datawith each other via the audio processing circuitry 3601. The audioprocessing circuitry 3601 may therefore comprise at least two interfacesfor device systems with a digital only path between interfaces.

The audio circuitry 3601 may also transmit data to external devices,such as accessory or peripheral apparatuses. For instance the audiocircuitry 3601 may communicate via suitable connections with outputtransducers 3610 and/or receive data from external input transducers3611.

Referring back to FIG. 35 such external apparatuses may be connected invarious ways. For instance a headset comprising earphones 3507 forreceiving stereo data and a voice microphone 3508 for receiving voiceaudio may be connected via a suitable jack plug to a headphone socket ofthe device. The same socket could also be used for driving line-outsignals to an external audio device, or such an external audio device3509 comprising speakers 3510 and an amplifier unit 3511 could beconnected via a different connector, possibly via a docking station.Wireless communication may also be used to communicate audio datawirelessly to a suitable device, such as wireless headset 3512 havingspeakers 3513 and microphone 3514. As shown in FIG. 36 wirelesscommunication may occur via wireless transceiver 3608.

Embodiments described above have assumed that all incoming PDM datastreams have the same sample rate. If say one data stream has say halfthe sample rate, then each sample may obviously be sampled twice insuccessive clock intervals to provide an equivalent system. Similarlyeach sample of a PDM output stream may be sampled more than once toprovide a higher effective sample rate.

Embodiments described above receive one-bit, i.e. two-level, streams ofinput data. The invention may be adapted to process multi-level inputs,e.g. 3-level inputs (−1, 0, +1) as used to directly drive Class DH-bridges for example, by suitable adaptation of the truth tables, andsimilarly to output multi-level output pulse streams.

It can therefore be seen that embodiments of the present invention canbe used to provide efficient data transfer using only a limited numberof connections and which offers the ability to send additional data whenrequired and/or reduce any EMI effects by making use of alternative dataencodings for combinations of input data.

The embodiments herein have been described in relation to audio data.However, although reference is made herein to “audio data”, theelectrical signals that are handled by the circuitry can represent anyphysical phenomenon. For example, the term “audio data” can mean notjust signals that represent sounds that are audible by the human ear(for example in the frequency range of 20 Hz-20 kHz), but can also meaninput and/or output signals to and/or from haptic transducers (typicallyat frequencies below 20 Hz, or at least below 300 Hz) and/or inputand/or output signals to and/or from ultrasonic transducers (for examplein the frequency range of 20 kHz-300 kHz) and/or to infrasonictransducers (typically at frequencies below 20 Hz).

Therefore, it will be appreciated that the principles disclosed hereinmay be applied to other types of data for driving transducers such ashaptic transducers, ultrasonic transducers, hearing aid coils and thelike or for receiving data from transducers. In general for a givenfunctional unit of a device there may be various consumers of datatransmitted from that unit and various generators of data to be receivedby that functional unit, as described above in the context of an audioDSP/codec. The embodiments of the present invention are equallyapplicable to other types of data that may be sent from a functionalunit to consumers and/or sent to the functional unit from generators.

It will be appreciated that the interface circuitry may conveniently beimplemented, at least partly, as an integrated circuit and may form partof a host electronic device, especially a portable device and/or abattery powered device. It will further be appreciated that the digitaldata may be manipulated by either electronic circuitry or in software orin a combination of both. The interface circuitry may be used in anaudio device such as a personal music or video player. The amplifier maybe implemented in a mobile communications device such as mobiletelephone or a computing device, such as a laptop or tablet computer orPDA. The interface circuitry may be used in a gaming device.

It should be noted that the above-mentioned embodiments illustraterather than limit the invention, and that those skilled in the art willbe able to design many alternative embodiments without departing fromthe scope of the appended claims. The word “comprising” does not excludethe presence of elements or steps other than those listed in a claim,“a” or “an” does not exclude a plurality, and a single feature or otherunit may fulfil the functions of several units recited in the claims.The word “amplify” can also mean “attenuate”, i.e. decrease, as well asincrease and vice versa and the word “add” can also mean “subtract”,i.e. decrease, as well as increase and vice versa. Any referencenumerals or labels in the claims shall not be construed so as to limittheir scope.

The invention claimed is:
 1. An audio accessory apparatus for playbackof audio signals received, in use, from an electronic device comprising:at least first and second audio output transducers; a connector forremovably connecting the audio accessory apparatus to the electronicdevice in use, the connector having a plurality of contacts; andinterface circuitry for receiving said audio signals via said connector,the interface circuitry comprising a digital module for receivingdigital audio signals via said connector and driving said audio outputtransducers based on said received digital audio signals; wherein theinterface circuitry is operable: in a first mode to connect first andsecond contacts of said plurality of contacts directly to the first andsecond audio output transducers respectively to drive said audio outputtransducers with analogue audio signals received at said first andsecond contacts; and in a second mode to disconnect the first and secondcontacts of the plurality of contacts from the first and second audiooutput transducers respectively and to provide at least a first digitalsignal path from the first contact to the digital module; wherein, inthe second mode of operation, the interface circuitry is configured toderive power for said digital module from at least one power supplyingcontact of said plurality of contacts of said connector; and wherein theinterface circuitry comprises discrimination circuitry configured to,when power is available via the at least one power supplying contact,determine whether to operate in the second mode and, if so, generate acontrol signal to disable the analogue signal paths.
 2. An audioaccessory apparatus as claimed in claim 1 comprising first and secondswitches for enabling/disabling the first and second analogue signalpaths respectively wherein the first and second switches default toenabling the first and second analogue signal paths.
 3. An audioaccessory apparatus as claimed in claim 1 wherein, in the second mode ofoperation, the interface circuitry is configured to derive power forsaid digital module from at least one power supplying contact of saidplurality of contacts of said connector.
 4. An audio accessory apparatusas claimed in claim 3 wherein said at least one power supplying contactcomprises at least one of the first and second contacts.
 5. An audioaccessory apparatus as claimed in claim 3 wherein, in the second mode ofoperation, one of said at least one power supplying contacts is used toreceive a digital audio signal and the interface comprises a powerharvesting module for deriving power from the received digital audiosignal.
 6. An audio accessory apparatus as claimed in claim 1 whereinsaid discrimination circuitry is configured to, when power is availablevia the at least one power supplying contact, to attempt handshakingwith a device connected via said connector.
 7. An audio accessoryapparatus as claimed in claim 1 further comprising a first microphone,wherein said connector comprises a third contact and the interfacecircuitry is operable in the first mode to provide a third analoguesignal path for providing a microphone bias from the third contact tothe first microphone.
 8. An audio accessory apparatus as claimed inclaim 7 wherein the interface circuitry is operable in the second modeof operation to operate the microphone and derive a digital microphonesignal and to transmit the digital microphone signal to the connectedelectronic device via said connector.
 9. An audio accessory apparatus asclaimed in claim 7 wherein in the second mode of operation the interfacecircuitry is configured to provide a second digital signal path from oneof the second and third contacts to the digital module, wherein one ofsaid first and second digital signal paths is used to receive digitaldata from the electronic device and the other of said first and seconddigital signal paths is used to transmit digital data to the electronicdevice.
 10. An audio accessory apparatus as claimed in claim 1 furthercomprising at least one additional audio transducer wherein theapparatus is configured such the at least one additional audiotransducer is disabled in the first operating mode and enabled in thesecond operating mode.
 11. An audio accessory apparatus as claimed inclaim 10 wherein the at least one additional audio transducer comprisesat least one microphone.
 12. An audio accessory apparatus as claimed inclaim 1 wherein said connector is a jack plug.
 13. An audio accessoryapparatus as claimed in claim 12 wherein said connector is a TRS jackplug or a TRRS jack plug.
 14. Audio interface circuitry for an accessoryapparatus which can be removably connected to an electronic device via aremovable connector, the audio interface circuitry comprising: aplurality of connector nodes for coupling to respective contacts of theremovable connector; and a first output node for outputting analogueaudio signals for driving an audio output transducer; wherein the audiointerface circuitry is operable: in an analogue mode to forward analogueaudio signals received at a first connector node of said plurality ofconnector nodes to the first output node via an analogue signal path;and in a digital mode to disable the analogue signal path and forwarddigital audio signals received at the first connector node to a digitaldecoder, wherein the digital decoder converts the digital audio signalto an analogue audio signal which is forwarded to the first output node;wherein in the digital mode the digital decoder is configured to receivean operating power supply from one of said plurality of connector nodesand in the absence of suitable power from the one of said plurality ofconnector nodes the audio interface circuitry is configured to defaultto the analogue mode; and wherein the audio interface circuitrycomprises discrimination circuitry configured to, when power isavailable via the one of said plurality of connector nodes, determinewhether to operate in the digital mode and, if so, generate a controlsignal to disable the analogue signal path.